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# R.M.

D ENGINEERING COLLEGE
DEPARTMENT OF ECE
QUESTION BANK
DIGITAL SIGNAL PROCESSING

BRANCH/SEM/SEC:CSE/IV/A& B

UNIT I

SIGNALS AND SYSTEMS

Part – A

1. What do you understand by the terms : signal and signal processing
2. Determine which of the following signals are periodic and compute their
fundamental period (AU DEC 07)
a) sin√2 Лt b)sin20 Лt+sin5Лt
3. What are energy and power signals? (MU Oct. 96)
4. State the convolution property of Z transform (AU DEC 06)
5. Test the following systems for time invariance: (DEC 03)
a) y(n)=n x
2
(n) b)a
x(n)

6. Define symmetric and antisymmetric signals. How do you prevent alaising while
sampling a CT signal? (AU MAY 07)(EC 333, May „07)
7. What are the properties of region of convergence(ROC) ?(AU MAY 07)
8. Differentiate between recursive and non recursive difference equations
(AU APR 05)
9. What are the different types of signal representation?
10. Define correlation (AU DEC 04)
11. what is the causality condition for LTI systems? (AU DEC 04)
12. define linear convolution of two DT signals (AU APR 04)
13. Define system function and stability of DT system (AU APR 04)
14. Define the following (a) System (b) Discrete-time system
15. What are the classifications of discrete-time system?
16. What is the property of shift-invariant system?
17. Define (a) Static system (b) Dynamic system? (AU DEC 03)
18. define cumulative and associative law of convolution (AU DEC 03)
19. Define a stable and causal system
20. What is the necessary and sufficient condition on the impulse response for
stability? (MU APR.96)
21. What do you understand by linear convolution? (MU APR. 2000)
22. What are the properties of convolution? (AU IT Dec. 03)
23. State Parseval‟s energy theorem for discrete-time aperiodic signals(AU DEC 04)
24. Define DTFT pair (MU Apr. 99)
25. What is aliasing effect? (AU MAY 07) (EC 333 DEC 03)
26. State sampling theorem
27. What is an anti-aliasing filter?
28. What is the necessary and sufficient condition on the impulse response for
stability? (EC 333, May „07)
22. State the condition for a digital filter to be causal and stable

Part-B

1. a) Compute the convolution y(n) of the signals (AU DEC 07)
x(n)= a
n
-3≤n≤5
0 elsewhere and
h(n)= 1 0≤n≤4
0 elsewhere

b) A discrete time system can be static or dynamic, linear or non-linear, Time
variant or time invariant, causal or non causal, stable or unstable. Examine the
following system with respect to the properties also (AU DEC 07)
1) y(n)=cos(x(n))
2) y(n)=x(-n+2)
3) y(n)=x(2n)
4)y(n)=x(n) cos ωn

2. a) Determine the response of the causal system
y(n)-y(n-1)=x(n)+x(n-1) to inputs x(n)=u(n) and x(n)=2
–n
u(n).Test its stability
b) Determine the IZT of X(Z)=1/(1-Z
-1
)(1-Z
-1
)
2
(AU DEC 07)

Determine whether each of the following systems defined below is (i) casual (ii)
linear (iii) dynamic (iv) time invariant (v) stable
(a) y(n) = log
10
[{x(n)}]
(b) y(n) = x(-n-2)
(c) y(n) = cosh[nx(n) + x(n-1)]

3. Compute the convolution of the following signals
x(n) = {1,0,2,5,4} h(n) = {1,-1,1,-1}
↑ ↑

h(n) = {1,0,1} x(n) = {1,-2,-2,3,4}
↑ ↑

4. Find the convolution of the two signals
x(n) = 3
n
u(-n); h(n) = (1/3)
n
u(n-2)
x(n) = (1/3)
–n
u(-n-1); h(n) = u(n-1)
x(n) = u(n) –u(n-5); h(n) = 2[u(n) – u(n-3)]

5. Find the discrete-time Fourier transform of the following
x(n) = 2
-2n
for all n
x(n) = 2
n
u(-n)
x(n) = n [1/2] (n)

6. Determine and sketch the magnitude and phase response of the following systems
(a) y(n) = 1/3 [x(n) + x(n-1) + x(n-2)]
(b) y(n) = ½[x(n) – x(n-1)]
(c) y(n) - 1/2y(n-1)=x(n)

7. a) Determine the impulse response of the filter defined by y(n)=x(n)+by(n-1)
b) A system has unit sample response h(n) given by
h(n)=-1/δ(n+1)+1/2δ(n)-1-1/4 δ(n-1). Is the system BIBO stable? Is the filter

8. Determine the Fourier transform of the following two signals(CS 331 DEC 2003)
a) a
n
u(n) for a<1
b) cos ωn u(n)

9. Check whether the following systems are linear or not (AU APR 05)
a) y(n)=x
2
(n) b) y(n)=n x(n)

10. For each impulse response listed below, dtermine if the corresponding system is
i) causal ii) stable (AU MAY 07)
1) 2
n
u(-n)
2) sin nЛ/2 (AU DEC 04)
3) δ(n)+sin nЛ
4) e
2n
u(n-1)
11. Explain with suitable block diagram in detail about the analog to digital
conversion and to reconstruct the analog signal (AU DEC 07)

12. Find the cross correlation of two sequences
x(n)={1,2,1,1} y(n)={1,1,2,1} (AU DEC 04)

13. Determine whether the following systems are linear , time invariant
1) y(n)=A x(n)+B
2) y(n)=x(2n)
Find the convolution of the following sequences: (AU DEC 04)
1) x(n)=u(n) h(n)=u(n-3)
2) x(n)={1,2,-1,1} h(n)={1,0,1,1}

UNIT II

FAST FOURIER TRANSFORMS

1) THE DISCRETE FOURIER TRANSFORM

PART A

1. Find the N-point DFT of a sequence x(n) ={1 ,1, 2, 2}
2. Determine the circular convolution of the sequence x1(n)={1,2,3,1} and
x2(n)={4,3,2,1} (AU DEC 07)
3. Draw the basic butterfly diagram for radix 2 DIT-FFT and DIF-FFT(AU DEC 07)
4. Determine the DTFT of the sequence x(n)=a
n
u(n) for a<1 (AU DEC 06)
5. Is the DFT of the finite length sequence periodic? If so state the reason
(AU DEC 05)
6. Find the N-point IDFT of a sequence X(k) ={1 ,0 ,0 ,0} (Oct 98)
7. what do you mean by „in place‟ computation of FFT? (AU DEC 05)
8. What is zero padding? What are its uses? (AU DEC 04)
9. List out the properties of DFT (MU Oct 95,98,Apr 2000)
10. Compute the DFT of x(n)=∂(n-n
0
)
11. Find the DFT of the sequence of x(n)= cos (n∏/4) for 0≤n≥ 3 (MU Oct 98)
12. Compute the DFT of the sequence whose values for one period is given by
x(n)={1,1,-2,-2}. (AU Nov 06,MU Apr 99)
13. Find the IDFT of Y(k)={1,0,1,0} (MU Oct 98)
14. What is zero padding? What are its uses?
15. Define discrete Fourier series.
16. Define circular convolution
17. Distinguish between linear convolution and Circular Convolution.
(MU Oct 96,Oct 97,Oct 98)
18. Obtain the circular convolution of the following sequences x(n)={1, 2, 1} and
h(n)={1, -2, 2}
19. Distinguish between DFT and DTFT (AU APR 04)
20. Write the analysis and synthesis equation of DFT (AU DEC 03)
21. Assume two finite duration sequences x
1
(n) and x
2
(n) are linearly combined.
What is the DFT of x3(n)?(x3(n)=Ax1(n)+Bx2(n)) (MU Oct 95)
22. If X(k) is a DFT of a sequence x(n) then what is the DFT of real part of x(n)?
23. Calculate the DFT of a sequence x(n)=(1/4)^n for N=16 (MU Oct 97)
24. State and prove time shifting property of DFT (MU Oct 98)
25. Establish the relation between DFT and Z transform (MU Oct 98,Apr 99,Oct 00)
26. What do you understand by Periodic convolution? (MU Oct 00)
27. How the circular convolution is obtained using concentric circle method?
(MU Apr 98)
28. State the circular time shifting and circular frequency shifting properties of DFT
29. State and prove Parseval‟s theorem
30. Find the circular convolution of the two sequences using matrix method
X1(n)={1, 2, 3, 4} and x2(n)={1, 1, 1, 1}
31. State the time reversal property of DFT
32. If the DFT of x(n) is X(k) then what is the DFT of x*(n)?
33. State circular convolution and circular correlation properties of DFT
34. Find the circular convolution of the following two sequences using concentric
circle method
x
1
(n)={1, 2, 3, 4} and x
2
(n)={1, 1, 1, 1}
35. The first five coefficients of X(K)={1, 0.2+5j, 2+3j, 2 ,5 }Find the remaining
coefficients

PART B

1. Find 4-point DFT of the following sequences
(a) x(n)={1,-1,0,0}
(b) x(n)={1,1,-2,-2} (AU DEC 06)
(c) x(n)=2
n

(d) x(n)=sin(n∏/2)

2. Find 8-point DFT of the following sequences
(a) x(n)={1,1,1,1,0,0,0,0}
(b) x(n)={1,2,1,2}

3. Determine IDFT of the following
(a)X(k)={1,1-j2,-1,1+j2}
(b)X(k)={1,0,1,0}
(c)X(k)={1,-2-j,0,-2+j}

4. Find the circular convolution of the following using matrix method and
concentric circle method
(a) x
1
(n)={1,-1,2,3}; x
2
(n)={1,1,1};
(b) x
1
(n)={2,3,-1,2}; x
2
(n)={-1,2,-1,2};
(c) x
1
(n)=sin n∏/2; x
2
(n)=3
n
0≤n≥7

5.Calculate the DFT of the sequence x(n)={1,1,-2,-2}
Determine the response of the LTI system by radix2 DIT-FFT? (AU Nov 06).
If the impulse response of a LTI system is h(n)=(1,2,3,-1)

6. Determine the impulse response for the cascade of two LTI systems having
impulse responses h
1
(n)=(1/2)^n* u(n),h
2
(n)=(1/4)^n*u(n) (AU May 07)

7. Determine the circular convolution of the two sequences x
1
(n)={1, 2, 3, 4}
x
2
(n)={1, 1, 1, 1} and prove that it is equal to the linear convolution of the same.

8. Find the output sequence y(n)if h(n)={1,1,1,1} and x(n)={1,2,3,1} using circular
convolution (AU APR 04)

9. State and prove the following properties of DFT (AU DEC 03)
1) Cirular convolution 2) Parseval‟s relation
2) Find the circular convolution of x1(n)={1,2,3,4} x2(n)={4,3,2,1}

2) FAST FOURIER TRANSFORM

PART A

1. Why FFT is needed? (AU DEC 03) (MU Oct 95,Apr 98)
2. What is FFT? (AU DEC 06)
3. Obtain the block diagram representation of the FIR filter (AU DEC 06)
4. Calculate the number of multiplications needed in the calculation of DFT and FFT
with 64 point sequence. (MU Oct 97, 98).
5. What is the main advantage of FFT?
6. What is FFT? (AU Nov 06)
7. How many multiplications and additions are required to compute N-point DFT
using radix 2 FFT? (AU DEC 04)
8. Draw the direct form realization of FIR system (AU DEC 04)
9. What is decimation-in-time algorithm? (MU Oct 95).
10. What do you mean by „in place‟ computation in DIT-FFT algorithm?
(AU APR 04)
11. What is decimation-in-frequency algorithm? (MU Oct 95,Apr 98).
12. Mention the advantage of direct and cascade structures (AU APR 04)
13. Draw the direct form realization of the system y(n)=0.5x(n)+0.9y(n-1)
(AU APR 05)
14. Draw the flow graph of a two point DFT for a DIT decomposition.
15. Draw the basic butterfly diagram for DIT and DIF algorithm. (AU 07).
16. How do we can calculate IDFT using FFT algorithm?
17. What are the applications of FFT algorithms?
18. Find the DFT of sequence x(n)={1,2,3,0} using DIT-FFT algorithms
19. Find the DFT of sequence x(n)={1,1, 1, 1} using DIF-FFT algorithms
(AU DEC 04)

PART B

1. Compute an 8-point DFT of the following sequences using DIT and DIF
algorithms
(a)x(n)={1,-1,1,-1,0,0,0,0}
(b)x(n)={1,1,1,1,1,1,1,1} (AU APR 05)
(c)x(n)={0.5,0,0.5,0,0.5,0,0.5,0}
(d)x(n)={1,2,3,2,1,2,3,2}
(e)x(n)={0,0,1,1,1,1,0,0} (AU APR 04)

2. Compute the 8 point DFT of the sequence x(n)={0.5, 0.5 ,0.5,0.5,0,0,0,0} using
radix 2 DIF and DIT algorithm (AU DEC 07)

3. a) Discuss the properties of DFT
b) Discuss the use of FFT algorithm in linear filtering (AU DEC 07)

4. How do you linear filtering by FFT using save-add method (AU DEC 06)

5. Compute the IDFT of the following sequences using (a)DIT algorithm (b)DIF
algorithms
(a)X(k)={1,1+j,1-j2,1,0,1+j2,1+j}
(b)X(k)={12,0,0,0,4,0,0,0}
(c)X(k)={5,0,1-j,0,1,0,1+j,0}
(d)X(k)={8,1+j2,1-j,0,1,0,1+j,1-j2}
(e)X(k)={16,1-j4.4142,0,1+j0.4142,0,1-j0.4142,0,1+j4.4142}

6. Derive the equation for DIT algorithm of FFT.
How do you do linear filtering by FFT using Save Add method? (AU Nov 06)

7. a) From first principles obtain the signal flow graph for computing 8 point DFT

b) Using the above signal flow graph compute DFT of x(n)=cos(n*Л)/4 ,0<=n<=7
(AU May 07).

8. Draw the butterfly diagram using 8 pt DIT-FFT for the following sequences
x(n)={1,0,0,0,0,0,0,0} (AU May 07).

9. a) From first principles obtain the signal flow graph for computing 8 point DFT
b) Using the above signal flow graph compute DFT of x(n)=cos(n*Л)/4 ,0<=n<=7

10. State and prove circular time shift and circular frequency shift properties of DFT

11. State and prove circular convolution and circular conjugate properties of DFT

12. Explain the use of FFT algorithms in linear filtering and correlation

13. Determine the direct form realization of the following system
y(n)=-0.1y(n-1)+0.72y(n-2)+0.7x(n)-0.252x(n-2) (AU APR 05)

14. Determine the cascade and parallel form realization of the following system
y(n)=-0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2)
Expalin in detail about the round off errors in digital filters (AU DEC 04)

UNIT-III

IIR FILTER DESIGN

PART-A

1. Distinguish between Butterworth and Chebyshev filter
2. What is prewarping? (AU DEC 03)
3. Distinguish between FIR and IIR filters (AU DEC 07)
4. Give any two properties of Butterworth and chebyshev filters (AU DEC 06)
5. Give the bilinear transformation (AU DEC 03)
6. Determine the order of the analog butterworth filter that has a -2 dB pass band
attenuation at a frequency of 20 rad/sec and atleast -10 dB stop band attenuation at 30
7. By impulse invariant method obtain the digital filter transfer function and
differential equation of the analog filter H(S)=1/S+1 (AU DEC 07)
8. Give the expression for location of poles of normalized butterworth filter
(EC 333, May „07)
9. What are the parameters(specifications) of a chebyshev filter (EC 333, May „07)
10. Why impulse invariance method is not preferred in the design of IIR filter other than low
pass filter?
11. What are the advantages and disadvantages of bilinear transformation?(AU DEC 04)
12. Write down the transfer function of the first order butterworth filter having low pass
behavior (AU APR 05)

13. What is warping effect? What is its effect on magnitude and phase response?
14. Find the digital filter transfer function H(Z) by using impulse invariance method for the
analog transfer function H(S)= 1/S+2 (MAY AU ‟07)
15. Find the digital filter transfer function H(Z) by using bilinear transformation method for
the analog transfer function H(S)= 1/S+3
16. Give the equation for converting a normalized LPF into a BPF with cutoff frequencies O
l
and

O
u

17. Give the magnitude function of Butterworth filter. What is the effect of varying order of
N on magnitude and phase response?
18. Give any two properties of Butterworth low pass filters. (MU NOV 06).
19. What are the properties of Chebyshev filter? (AU NOV 06).
20. Give the equation for the order of N and cut off frequency O
c
of Butterworth filter.
21. Give the Chebyshev filter transfer function and its magnitude response.
22. Distinguish between the frequency response of Chebyshev Type I filter for N odd and N
even.
23. Distinguish between the frequency response of Chebyshev Type I & Type II filter.
24. Give the Butterworth filter transfer function and its magnitude characteristics for
different order of filters.
25. Give the equations for the order N, major, minor and axis of an ellipse in case of
Chebyshev filter.
26. What are the parameters that can be obtained from the Chebyshev filter specification?
(AU MAY 07).
27. Give the expression for the location of poles and zeros of a Chebyshev Type II filter.
28. Give the expression for location of poles for a Chebyshev Type I filter. (AU MAY 07)
29. Distinguish between Butterworth and Chebyshev Type I filter.
30. How one can design Digital filters from Analog filters.
31. Mention any two procedures for digitizing the transfer function of an analog filter.
(AU APR 04)
32. What are properties that are maintained same in the transfer of analog filter into a digital
filter.
33. What is the mapping procedure between s-plane and z-plane in the method of mapping of
differentials? What is its characteristics?
34. What is mean by Impulse invariant method of designing IIR filter?
35. What are the different types of structures for the realization of IIR systems?
36. Write short notes on prewarping.
38. What is warping effect? What is its effect on magnitude and phase response?
39. What is Bilinear Transformation?
40. How many numbers of additions, multiplications and memory locations are required to
realize a system H(z) having M zeros and N poles in direct form-I and direct form –II
realization?
41. Define signal flow graph.
42. What is the transposition theorem and transposed structure?
43. Draw the parallel form structure of IIR filter.
44. Give the transposed direct form –II structure of IIR second order system.
45. What are the different types of filters based on impulse response? (AU 07)
46. What is the most general form of IIR filter?

PART B

1. a) Derive bilinear transformation for an analog filter with system function
H(S)=b/S+a (AU DEC 07)
b) Design a single pole low pass digital IIR filter with-3 Db bandwidth of 0.2Л by using
bilinear transformation
2. a) Obtain the direct form I, Direct form II, cascade and parallel realization for the
following
Systems
y(n)=-0.1x(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2)
b) Discuss the limitation of designing an IIR filetr using impulse invariant method
(AU DEC 07)

3. Determine H(Z) for a Butterworth filter satisfying the following specifications:
0.8 s |H(e
je
(s 1, for 0s es t/4
|H(e
je
(s 0.2, for t/2s es t
Assume T= 0.1 sec. Apply bilinear transformation method (AU MAY 07)

4.Determine digital Butterworth filter satisfying the following specifications:
0.707 s |H(e
je
(s 1, for 0s es t/2
| H(e
je
(s 0.2, for3t/4s es t
Assume T= 1 sec. Apply bilinear transformation method. Realize the filter in mose
convenient form (AU DEC 06)

5. Design a Chebyshev lowpass filter with the specifications o
p
=1 dB ripple in the pass
band 0ses0.2t, o
s
=15 dB ripple in the stop band 0.3 tsest using impulse invariance
method(AU DEC 06)

6. Design a Butterworth high pass filter satisfying the following specifications.
o
p
=1 dB; o
s
=15 dB
O
p
=0.4H; O
s
=0.2H

7. Design a Butterworth low pass filter satisfying the following specifications.
(AU DEC 04)
]
p
=0.10 Hz;o
p
=0.5 dB
]
s
=0.15 HZ;o
s
=15 dB:F=1Hz.

8. Design (a) a Butterworth and (b) a Chebyshev analog high pass filter that will
pass all radian frequencies greater than 200 rad/sec with no more that 2 dB
attuenuation and have a stopband attenuation of greater than 20 dB for all O less

9. Design a digital filter equivalent to this using impulse invariant method
H(S)=10/S
2
+7S+10 (AU DEC 03)(AU DEC 04)

10. Use impulse invariance to obtain H(Z) if T= 1 sec and H(s) is
1/(s
3
+3s
2
+4s+1)
1/(s
2
+\2 s +1)

11. Use bilinear transformation method to obtain H(Z) if T= 1 sec and H(s) is
1/(s+1)(S+2) (AU DEC 03)
1/(s
2
+\2 s +1)

12. Briefly explain about bilinear transformation of digital filter design(AU APR 05)

13. Use bilinear transform to design a butterworth LPF with 3 dB cutoff frequeny of
0.2H (AU APR 04)

14. Compare bilinear transformation and impulse invariant mapping

15. a) Design a chebyshev filter with a maxmimum pass band attenuation of 2.5 Db;
at Ωp=20 rad/sec and the stop band attenuation of 30 Db at Ωs=50 rad/sec.
b)Realize the system given by difference equation
y(n)=-0.1 y(n-1)+0.72y(n-2)+0.7x(n)-0.25x(n-2) in parallel form
(EC 333 DEC „07 )

UNIT IV

FIR FILTER DESIGN

PART A

1. What are the desirable and undesirable features of FIR filter?
2. Discuss the stability of the FIR filters (AU APR 04) (AU DEC 03)
3. What are the main advantages of FIR over IIR (AU APR 04)
4. What is the condition satisfied by Linear phase FIR filter? (DEC 04) (EC 333
MAY 07)
5. What are the design techniques of designing FIR filters?
6. What condition on the FIR sequence h(n) are to be imposed in order that this filter can be
called a Linear phase filter? (AU 07)
7. State the condition for a digital filter to be a causal and stable. (AU 06)
8. What is Gibbs phenomenon? (AU DEC 04) (AU DEC 07)
9. Show that the filter with h(n)={-1, 0, 1} is a linear phase filter
10. Explain the procedure for designing FIR filters using windows. (MU 02)
11. What are desirable characteristics of windows?
12. What is the principle of designing FIR filters using windows?
13. What is a window and why it is necessary?
14. Draw the frequency response of N point rectangular window. (MU 03)
15. Give the equation specifying Hanning and Blackman windows.
16. Give the expression for the frequency response of
17. Draw the frequency response of N point Bartlett window
18. Draw the frequency response of N point Blackman window
19. Draw the frequency response of N point Hanning window. (AU DEC 03)
20. What is the necessary and sufficient condition for linear phase characteristics in FIR
filter. (MU Nov 03)
21. Give the equation specifying Kaiser window.
22. Compare rectangular and hanning window functions
23. Briefly explain the frequency sampling method of filter design
24. Compare frequency sampling and windowing method of filter design

PART-B

1. Use window method with a Hamming window to design a 13-tap differentiator
(N=13). (AU „07)

2. i) Prove that FIR filter has linear phase if the unit impulse responsesatisfies the
condition h(n)=h(N-1-n), n=0,1,……M-1. Also discuss symmetric and
antisymmetric cases of FIR filter (AU DEC 07)

3. What are the issues in designing FIR filter using window method?(AU APR 04,
DEC 03)
4. ii) Explain the need for the use of window sequences in the design of FIR filter.
Describe the window sequences generally used and compare their properties

5. Derive the frequency response of a linear phase FIR filter when impulse responses
symmetric & order N is EVEN and mention its applications

6. i) Explain the type I design of FIR filter using frequency sampling method
ii) A low pass filter has the desired response as given below
Hd(e
je
)= e –j3e, 0≤e≤Л/2
0 Л/2≤e≤Л
Determine the filter coefficients h(n) for M=7 using frequency sampling
technique (AU DEC 07)

7. i) Derive the frequency response of a linear phase FIR filter when impulse responses
antisymmetric & order N is odd
ii) Explain design of FIR filter by frequency sampling technique (AU MAY 07)

7. Design an approximation to an ideal bandpass filter with magnitude response
H(e
je
) = 1 ; [/4s,e,s3[/4
0 ; otherwise
Take N=11. (AU DEC 04)

8. Design a 15-tap linear phase filter to the following discrete frequency response
(N=15) using frequency sampling method (MU 03)
H(k) = 1 0 s k s 4
= 0.5 k=5
= 0.25 k=6
= 0.1 k=7
= 0 elsewhere

9. Design an ideal band pass digital FIR filter with desired frequency response
H(e
je
)= 1 for 0.25t s |e(s 0.75t
0 for(e(s 0.25t and 0.75 ts|e(st
by using rectangular window function of length N=11. (AU DEC 07)

10. Design an Ideal Hilbert transformer using hanning window and
Blackman window for N=11. Plot the frequency response in both
Cases

11. a) How is the design of linear phase FIR filter done by frequency sampling method?
Explain.
b) Determine the coefficients of a linear phase FIR filter of length N=15 which has
Symmetric unit sample response and a frequency response that satisfies the following
conditions

H
r
(2t k/15) = 1 for k=0,1,2,3
0 for k=4
0 for k=5,6,7
12. An FIR filter is given by the difference equation
y(n)=2x(n)+4/5 x(n-1)+3/2 x(n-2)+2/3 x(n-3) Determine its lattice form(EC 333 DEC 07)

13. Using a rectangular window technique design a low pass filter with pass band gain of unity
cut off frequency of 1000 Hz and working at a sampling frequency of 5 KHz. The length
of the impulse response should be 7.( EC 333 DEC 07)

16. Design an Ideal Hilbert transformer using rectangular window and Black man window
for N=11. Plot the frequency response in both Cases (EC 333 DEC ‟07)

9. 17. Design an approximation to an ideal lowpass filter with magnitude response
H(e
je
) = 1 ; 0s,e,s[/4
0 ; otherwise
Take N=11.Use hanning and hamming window (AU DEC 04)

UNIT V

FINITE WORD LENGTH EFFECTS

PART –A

1. What do you understand by a fixed point number? (MU Oct‟95)
2. Express the fraction 7/8 and -7/8 in sign magnitude, 2‟s complement and 1‟s
complement (AU DEC 06)
3. What are the quantization errors due to finite word length registers in digital filters?
(AU DEC 06)
4. What are the different quantization methods? (AU DEC 07)
5. What are the different types of fixed point number representation?
6. What do you understand by sign-magnitude representation?
7. What do you understand by 2‟s complement representation?
8. Write an account on floating point arithmetic? (MU Apr 2000)
9. What is meant by block floating point representation? What are its advantages?
10. what are advantages of floating point arithmetic?
11. Compare the fixed point and floating point arithmetic. (MU Oct‟96)
12. What are the three quantization errors due to finite word length registers in
digital filters? (MU Oct‟98)
13. How the multiplication and addition are carried out in floating point
arithmetic?
14. Brief on co-efficient inaccuracy.
15. What do you understand by input quantization error?
16. What is product quantization error?
17. What is meant by A/D conversion mode?
18. What is the effect of quantization on pole locations?
19. What are the assumptions made concerning the statistical independence of
various noise sources that occur in realizing the filter? (M.U. Apr 96)
20. What is zero input limit cycle overflow oscillation (AU 07)
21. What is meant by limit cycle oscillations?(M.U Oct 97, 98, Apr 2000) (AU DEC 07)
29. Explain briefly the need for scaling digital filter implementation?
(M.U Oct 98)(AU-DEC 07)
30. Why rounding is preferred than truncation in realizing digital filter? (M.U. Apr 00)
31. Define the deadband of the filter? (AU 06)
25. Determine the dead band of the filter with pole at 0.5 and the number of bits used
for quantization is 4(including sign bit)
26. Draw the quantization noise model for a first order IIR system
27. What is meant by rounding? Draw the pdf of round off error
28. What is meant by truncation? Draw the pdf of round off error
29. What do you mean by quantization step size?
30. Find the quantization step size of the quantizer with 3 bits
31. Give the expression for signal to quantization noise ratio and calculate the
improvement with an increase of 2 bits to the existing bit.
32. Express the following binary numbers in decimal

A) (100111.1110)
2
(B) (101110.1111)
2
C (10011.011)
2
33.Why rounding is preferred to truncation in realizing digital filter? (EC 333, May „07)
34. List the different types of frequency domain coding (EC 333 MAY 07)
35. What is subband coding? (EC 333 MAY 07)

PART-B

1. Draw the quantization noise model for a second order system and explain
H(z)=1/(1-2rcosuz
-1
+r
2
z
-2
) and find its steady state output noise variance (ECE AU‟ 05)

2. Consider the transfer function H(z)=H
1
(z)H
2
(z) where
H
1
(z)=1/(1-a
1
z
-1
) , H
2
(z)=1/(1-a
2
z
-2
).Find the output round off noise power.
Assume a
1
=0.5 and a
2
=0.6 and find out the output round off noise power.
(ECE AU‟ 04)(EC 333 DEC 07)

3. Find the effect of coefficient quantiztion on pole locations of the given second
order IIR system when it is realized in direct form –I and in cascade form. Assume a
word length of 4-bits through truncation.
H(z)= 1/(1-0.9z
-1
+0.2z
–2
) (AU‟ Nov 05)

4. Explain the characteristics of Limit cycle oscillations with respect to the system described
by the differential equations.
y(n)=0.95y(n-1)+x(n) and
determine the dead band of the filter (AU‟ Nov 04)
5. i) Describe the quantization errors that occur in rounding and truncation in two‟s
complement
ii) Draw a sample/hold circuit and explain its operation
iii) What is a vocoder? Expalin with a block diagram (AU DEC 07)

6. Two first order low pass filter whose system functions are given below are connected in
cascade. Determine the overall output noise power
H1(Z)=1/(1-0.9Z
-1
) H2(Z)=1/(1-0.8Z
-1
) (AU DEC 07)

7. Consider a Butterworth lowpass filter whose transfer function is
H(z)=0.05( 1+z
-1
)
2
/(1-1.2z
-1
+0.8 z
-2
).
Compute the pole positions in z-plane and calculate the scale factor S
o
to prevent
8. Express the following decimal numbers in binary form
A) 525 B) 152.1875 C) 225.3275

10.

Express the decimal values 0.78125 and -0.1875 in
One‟s complement form
sign magnitude form
Two‟s complement form.

11.

Express the decimal values -6/8 and 9/8 in (i) Sign magnitude form (ii) One‟s complement
form (iii) Two‟s complement form

12.

Study the limit cycle behavior of the following systems
i.

y(n) = 0.7y(n-1) + x (n)
ii.

y(n) = 0.65y(n-2) + 0.52y (n-1) + x (n)
13.

For the system with system function H (z) =1+0.75z-1 / 1-0.4z-1 draw the signal flow graph
14.

and find scale factor s0 to prevent overflow limit cycle oscillations
15.

Derive the quantization input nose power and determine the signal to noise ratio of the system
16.

Derive the truncation error and round off error noise power and compare both errors
17.

Explain product quantization error and coefficient quantization error with examples
18.

Derive the scaling factor So that prevents the overflow limit cycle oscillations in a second
order IIR system.
19.

The input to the system y(n)=0.999y(n-1)+x(n) is applied to an ADC. What is the power
produced by the quantization noise at the output of the filter if the input is quantized to
1) 8 bits 2) 16 bits (EC 333 DEC 07)
19. Convert the following decimal numbers into binary: (EC 333 DEC 07)
1) (20.675)
10
2) (120.75)
10
20. Find the steady state variance of the noise in the output due to quantization of input for the
first order filter y(n)=ay(n-1)+x(n)
(EC 333 DEC 07)

ANAND INSTITUTE OF HIGHER TECHNOLOGY
KAZHIPATTUR, CHENNAI –603 103

DEPARTMENT OF ECE
Date: 15-05-2009
Subject : Digital signal Processing Sub Code : IT1252
Staff Name: Robert Theivadas.J Class : VII Sem/CSE A&B

UNIT-1 - SIGNALS AND SYSTEMS
PART A
1. Determine which of the following sinusoids are periodic and compute their fundamental
period
(a) Cos 0.01πn
(b) sin (π62n/10) Nov/Dec 2008 CSE

a) Cos 0.01 πn
Wo=0.01 π the fundamental frequency is multiply of π .Therefore the signal is
periodic
Fundamental period
N=2π [m/wo]
=2π(m/0.01π)
Choose the smallest value of m that will make N an integer
M=0.1
N=2π(0.1/0.01π)
N=20
Fundamental period N=20
b) sin (π62n/10)
Wo=0.01 π the fundamental frequency is multiply of π .Therefore the signal is
periodic
Fundamental period
N=2π [m/wo]
=2π(m/(π62/10))
Choose the smallest value of m that will make N an integer
M=31
N=2π(310/62π)
N=10
Fundamental period N=10
2. State sampling theorem Nov/Dec 2008 CSE
A band limited continuous time signal, with higher frequency f
max
Hz can be uniquely
recovered from its samples provided that the sampling rate Fs>2f
max
samples per second
3. State sampling theorem , and find Nyquist rate of the signal
x(t)=5 sin250 tt + 6cos300 tt April/May2008 CSE
A band limited continuous time signal, with higher frequency f
max
Hz can be
uniquely recovered from it‟s samples provided that the sampling rate Fs>2f
max
samples
per second.
Nyquist rate
x(t)=5 sin250tt+ 6cos300 tt
Frequency present in the signals
F1=125Hz F2=150Hz
Fmax=150Hz
Fs>2Fmax=300 Hz
The Nyquist rate is FN= 300Hz

4. State and prove convolution property of Z transform. April/May2008 CSE
Convolution Property (MAY 2006 ECESS)

5. Determine which of the following signals are periodic and compute their
fundamental period. Nov/Dec 2007 CSE
(a) sin √2пt
(b) sin 20пt + sin 5пt
(a) sin √2пt
wo=√2п .The Fundamental frequency is multiply of п.Therefore, the signal is
Periodic .
Fundamental period
N=2п [m/wo]
= 2п [m/√2п]
m=√2
=2п [√2/√2п]
N=2
(b) sin 20пt + sin 5пt
wo=20п, 5п .The Fundamental frequency is multiply of п.Therefore, the signal is
Periodic .
Fundamental period of signal sin 20пt
N1=2п [m/wo]
=2п [m/20п] m=1
=1/10
Fundamental period of signal sin 5пt
N2=2п [m/wo]
=2п [m/5п] m=1
=2/5
N1/N2=(1/10)/(2/5)
=1/4
4N1=N2
N= 4N1=N2
N=2/5
6. Determine the circular convolution of the sequence x1(n)={1,2,3,1} and
x2(n)={4,3,2,1}. Nov/Dec 2007 CSE
Soln:
x1(n)={1,2,3,1}
x2(n)={4,3,2,1}.

Y(n)= 15,16,21,15

7. Define Z transform for x(n)=-na
n
u(-n-1) April/May 2008 IT

X(n) =-na
n
u(-1-n)
X (z)=
= u(-n-1)=0for n>1
=
= -
= -z d/dz X(z)
=z d/dz( )=
8. Find whether the signal y= n
2
x(n) is linear April/May 2008 IT
Y= x(n)
Y1(n)=T[x1(n)]= x1(n)
Y2(n)= T[x2(n)]= x2(n)
The weighted sum of input is
a1 T[x1(n)]+a2 T[x2(n)]=a1 x1(n)+a2 x2(n)-----------1
the output due to weighted sum of input is
y3(n)=T[a1X1(n)+a2X2(n)]
= a1 x1(n)+a2 x2(n)----------------------------------2

9. Is the system y(n)=ln[x(n)] is linear and time invariant? (MAY 2006 IT)
The system y(n)=ln[x(n)] is non-linear and time invariant
alnx
1
(n)+blnx
2
(n) ≠ ln(ax
1
(n)+bx
2
(n)  Non-linear system
lnx (n)=lnx (n-n
0
)  Time invariant system
10. Write down the expression for discrete time unit impulse and
unit step function. (APR 2005 IT).
Discrete time unit impulse function
δ(n) =1, n=0
=0, n≠0
Discrete time step impulse function.
u(n) = 1, for n≥0
= 0 for n<0
11. List the properties of DT sinusoids. (NOV 2005 IT)
- DT sinusoid is periodic only if its frequency f is a rational number.
- DT sinusoid whose frequencies are separated by an integer multiple of 2π are
identical.
12. Determine the response a system with y(n)=x(n-1) for the input signal
x(n) = |n| for -3≤n≤3
= 0 otherwise (NOV 2005 IT)
x(n)= {3,2,1,0,1,2,3}

y(n) = x(n-1) ={3,2,1,0,1,2,3}
13. Define linear convolution of two DT signals. (APR 2004 IT)
y(n)=x(n)*h(n), * represent the convolution operator
y(n), x(n)&h(n), Output, Input and response of the system respectively.
14. Define system function and stability of a DT system. (APR 2004 IT)
H(z)=Y(z)/X(z)
H(z),Y(z) & X(z)z-transform of the system impulse, output and input respectively.
15. What is the causality condition for an LTI system? (NOV 2004 IT)
Conditions for the causality
h(n)=0 for n<0
16. What are the different methods of evaluating inverse z transform.
(NOV 2004 IT)
- Long division method
- Partial fraction expansion method
- Residue method
- Convolution method

UNIT-II - FAST FOURIER TRANSFORMS

1. Find out the DFT of the signal X(n)= (n) Nov/Dec 2008 CSE

X(n)={1,0,0,0}

X(k)={1,1,1,1}
2. What is meant by bit reversal and in place commutation as applied to FFT?
Nov/Dec 2008
CSE
"Bit reversal" is just what it sounds like: reversing the bits in a binary word from
left to write. Therefore the MSB's become LSB's and the LSB's become MSB's.The data
ordering required by radix-2 FFT's turns out to be in "bit reversed" order, so bit-reversed
indexes are used to combine FFT stages.

Input sample
index
Binary
Representation
Bit reversed
binary
Bit reversal
sample index
0 000 000 0
1 001 100 4
2 010 010 2
3 011 110 6
4 100 001 1
5 101 101 5
6 110 011 3
7 111 111 7

3. Draw radix 4 butterfly structure for (DIT) FFT algorithm
April/May2008 CSE

4. Find DFT for {1,0,0,1}. April/May2008 CSE /April/May
2008 IT

5. Draw the basic butterfly diagram for radix 2 DIT-FFT and DIF-FFT.
Nov/Dec
2007 CSE
Butterfly Structure for DIT FFT MAY 2006 ECESS
&(NOV 2006 ITSS)
The DIT structure can be expressed as a butterfly diagram

The DIF structure expressed as a butterfly diagram

6. What are the advantages of Bilinear mapping April/May 2008 IT
 Aliasing is avoided
 Mapping the S plane to the Z plane is one to one
 The closed left half of the S plane is mapped onto the unit disk of the Z plane
7. How may multiplication and addition is needed for radix-2 FFT? April/May 2008 IT
Number of complex addition is given by N
Number of complex multiplication is given by N/2
8. Define DTFT pair? (May/June 2007)-ECE
The DTFT pairs are (MAY 2006 IT)
X(k) = x(n)e
-j2πkn/N
X(n) = x(k)e
j2πkn/N

9. Define Complex Conjugate of DFT property. (May/Jun 2007)-ECE
DFT
If x(n)↔X(k) then
N
X*(n)↔(X*(-k))
N
= X*(N- K)
10.Differentiate between DIT and DIF FFT algorithms. (MAY 2006 IT)

S.No DIT FFT algorithm DIF FFT algorithm
1 Decimation in time FFT algorithm Decimation in frequency FFT
algorithm
2 Twiddle factor k=(Nt/2
m
) Twiddle factor k=(Nt/2
M-m+1
)
11.Give any two properties of DFT (APR 2004 IT SS)
Linearity : DFT [ax(n)+b y(n)]=a X(K)+bX(K)
Periodicity: x(n+N)=x(n) for all n
X(K+N)=X(K) for all n
12.What are the advantages of FFT algorithm over direct computation of DFT?
(May/June 2007)-ECE
The complex multiplication in the FFT algorithm is reduced by (N/2) log2N times.
Processing speed is very high compared to the direct computation of DFT.
13. What is FFT? (Nov/Dec 2006)-
ECE
The fast Fourier transform is an algorithm is used to calculate the DFT. It is based on
fundamental principal of decomposing the computation of DFT of a sequence of the length N in
to successively smaller discrete Fourier Transforms. The FFT algorithm provides speed increase
factor when compared with direct computation of the DFT.
14.Determine the DIFT of a sequence x(n) = a
n
u(n) (Nov/Dec 2006)-ECE

X(K) = x(n) e
j2πkn/N

The given sequence x(n) = a
n
u(n)

DTFT{x(n)} = x(n) e
j2πkn/N
= (a e
j2πk/N
)
n

Where a
n
= 1-a
n
/(1-a)
X(K) = (1 – a
N
e
j2πk
)/ (1-ae
j2πk/N
)
15. What do you mean by in place computation in FFT. (APR 2005 IT)
FFT algorithms, for computing the DFT when the size N is a power of 2 and when it
is a power of 4
16.Is the DFT is a finite length sequence periodic. Then state the reason (APR 2005
ITDSP)
DFT is a finite length sequence periodic.
N-1
X(e
je
)= Σ x(n) e
-jen

n =0
X(e
je
) is continuous & periodic in e, with period 2π.

UNIT-III - IIR FILTER DESIGN

1. What are the requirements for converting a stable analog filter into a stable digital filter?
Nov/Dec 2008 CSE
 The JΩ axis in the s plane should be map into the unit circle in the Z plane .thus there
will be a direct relationship between the two frequency variables in the two domains
 The left half plane of the s plane should be map into the inside of the unit circle in the z –
plane .thus the stable analog filter will be converted to a stable digital filter
2. Distinguish between the frequency response of chebyshev type I and Type II filter
Nov/Dec 2008 CSE

Type I chebyshev filter

Type II chebyshev filter

Type I chebyshev filters are all pole filters that exhibit equirpple behavior in the pass
band and monotonic in stop band .Type II chebyshev filters contain both poles and zeros
and exhibits a monotonic behavior in the pass band and an equiripple behavior in the stop
band
3. What is the need for prewraping in the design of IIR filter Nov/Dec 2008 CSE
The warping effect can be eliminated by prewarping the analog filter .This can be done
by finding prewarping analog frequencies using the formula
Ω = 2tan
-1
ΩT/2
4.Write frequency translation for BPF from LPF April/May2008 CSE
Low pass with cut – off frequency Ώ
C
to band –pass with lower cut-off frequency Ώ
1
and
higher cut-off frequency Ώ
2
:
S ------------- Ώ
C
(
s2
+ Ώ1 Ώ2) / s (Ώ
2
- Ώ
1
)
The system function of the high pass filter is then

H(s) = H
p
{ Ώ
C
(

s
2
+ Ώ1 Ώ2) / s (Ώ
2
- Ώ
1
)}
5.Compare Butterworth, Chebyshev filters April/May2008
CSE

Butter Worth Filter Chebyshev filters.
Magnitude response of Butterworth filter
decreases monotonically, as frequency
Magnitude response of chebyshev filter
exhibits ripple in pass band
increases from 0 ∞
Poles on the butter worth lies on the circle

Poles of the chebyshev filter lies on the
ellipse
6. Determine the order of the analog Butterworth filter that has a -2 db pass band
attenuation at a frequency of 20 rad/sec and atleast -10 db stop band attenuation at 30
Nov/Dec 2007CSE
α
p
α
s
= 10 dB; Ωs = 30 rad/sec

log√10
0.1 αs
-1/ 10
0.1 αp
-1
N≥
Log α
s/
α
p

log√10

-1/ 10
0.2
-1
N≥
Log 30
/
20

≥3.37
Rounding we get N=4
7. By Impulse Invariant method, obtain the digital filter transfer function
and differential equation of the analog filter H(s)=1 / (s+1) Nov/Dec 2007
CSE
H(s) =1/(s+1)
Using partial fraction
H(s) =A/(s+1)
= 1/(s-(-1)
Using impulse invariance method
H (z) =1/1-e
-T
z
-1

AssumeT=1sec
H(z)=1/1-e
-1
z
-1

H(z)=1/1-0.3678z
-1

8.Distinguish between FIR and IIR filters. Nov/Dec 2007 CSE

Sl.No IIR FIR
1 H(n) is infinite duration H(n) is finite duration
2 Poles as well as zeros are
present. Sometimes all pole
filters are also designed.
These are all zero filters.
3 These filters use feedback
from output. They are
recursive filters.
These filters do not use
feedback. They are nonrecursive.
4 Nonlinear phase response. Linear
phase is obtained if
H(z) = ±Z
-1
H(Z
-1
)
Linear phase response for
h(n) = ± h(m-1-n)
5 These filters are to be designed for
stability
These are inherently stable
filters
6 Number of
multiplication requirement is less.
More
7 More complexity of implementation Less complexity of implementation
8 Less memory is required More memory is requied
9 Design procedure is complication Less complicated
10 Design methods:
1. Bilinear Transform
2. Impulse invariance.
Design methods:
1. Windowing
2. Frequency sampling
11 Can be used where sharp
cutoff characteristics
with minimum order are required
Used where linear phase
characteristic is essential.
9.Define Parsevals relation April/May 2008 IT
If X1(n) and X2(n) are complex valued sequences ,then
= 1/2∏j
1. Many to one mapping.
2. linear frequency relationship between analog and its transformed digital frequency,
Aliasing
11.What is frequency warping? (MAY 2006 IT DSP)
The bilinear transform is a method of compressing the infinite, straight analog
frequency axis to a finite one long enough to wrap around the unit circle only once.
This is also sometimes called frequency warping. This introduces a distortion in the
frequency. This is undone by pre-warping the critical frequencies of the analog filter
(cutoff frequency, center frequency) such that when the analog filter is transformed
into the digital filter, the designed digital filter will meet the desired specifications.
12. Give any two properties of Butterworth filter and chebyshev filter. (Nov/Dec 2006)
a. The magnitude response of the Butterworth filter decreases monotonically as the
frequency increases (Ώ) from 0 to ∞.
b. The magnitude response of the Butterworth filter closely approximates the ideal
response as the order N increases.
c. The poles on the Butterworth filter lies on the circle.
d. The magnitude response of the chebyshev type-I filter exhibits ripple in the pass
band.
e. The poles of the Chebyshev type-I filter lies on an ellipse.

S = (2/T) (Z-1) (Z+1)
13.Find the transfer function for normalized Butterworth filter of order 1 by determining
the pole values. (MAY 2006 IT DSP)
Poles = 2N
N=1
Poles = 2
14..Differentiate between recursive and non-recursive difference equations.
(APR 2005 ITDSP)
The FIR system is a non-recursive system, described by the difference
equation M-1
y(n) = Σ b
k
x(n-k)
k=0
The IIR system is a non-recursive system, described by the difference
equation N M
y(n) = Σ b
k
x(n-k)- Σ a
k
y(n-k)
k=0 k=1
15.Find the order and poles of Butterworth LPF that has -3dB bandwidth of 500 Hz and an
attenuation of -40 dB at 1000 Hz. (NOV 2005 ITDSP)
α
p
= -3dB α
s
= -40dB Ω
s
p
=500*2π
The order of the filter N ≥(log(λ/ε))/(log(Ω
s

p
))
λ = (10
0.1αs
-1)
1/2
= 99.995

ε = (10
0.1αp
-1)
1/2
= 0.9976
N = (log(99.995/0.9976))/(log(2000π/1000π)) = 2/0.3 = 6.64
N ≥ 6.64 = 7
Poles=2N=14
16.What is impulse invariant mapping? What is its limitation? (Apr/May 2005)-ECE
The philosophy of this technique is to transform an analog prototype filter into an
IIR discrete time filter whose impulse response [h(n)] is a sampled version of the analog
filter‟s impulse response, multiplied by T.This procedure involves choosing the response of
the digital filter as an equi-spaced sampled version of the analog filter.
17.Give the bilinear transformation. (Nov/Dec 2003)-ECE
The bilinear transformation method overcomes the effect of aliasing that is
caused due to the analog frequency response containing components at or beyond the
nyquist frequency. The bilinear transform is a method of compressing the infinite,
straight analog frequency axis to a finite one long enough to wrap around the unit
circle only once.
ITDSP)

(i) The main advantage direct form-II structure realization is that the number of delay
elements is reduced by half. Hence, the system complexity drastically reduces the
number of memory elements .
(ii) Cascade structure realization, the system function is expressed as a product of
several sub system functions. Each sum system in the cascade structure is realized in
direct form-II. The order of each sub system may be two or three (depends) or more.
19. What is prewarping? (Nov/Dec 2003)-ECE
When bilinear transformation is applied, the discrete time frequency is related
continuous time frequency as,
Ω = 2tan
-1
ΩT/2
This equation shows that frequency relationship is highly nonlinear. It is also
called frequency warping. This effect can be nullified by applying prewarping. The
specifications of equivalent analog filter are obtained by following relationship,
Ω = 2/T tan ω/2
This is called prewarping relationship.
UNIT-IV - FIR FILTER DESIGN

1.What is gibb’s Phenomenon. April/May2008 CSE
The oscillatory behavior of the approximation XN(W) to the
function X(w) at a point of discontinuity of X(w) is called Gibb‟s Phenomenon
2.Write procedure for designing FIR filter using windows. April/May2008 CSE
1. Begin with the desired frequency response specification Hd(w)
2. Determine the corresponding unit sample response hd(n)
3. Indeed hd(n) is related to Hd(w) by the Fourier Transform relation.
3.What are Gibbs oscillations? Nov/Dec 2007
CSE
Oscillatory behavior observed when a square wave is reconstructed from finite
number of harmonics.
The unit cell of the square wave is given by

Its Fourier series representation is

4. Explain briefly the need for scaling in the digital filter realization Nov/Dec 2007
CSE
To prevent overflow, the signal level at certain points in the digital filters must be
scaled so that no overflow occur in the adder
5. What are the advantages of FIR filters? April/May 2008 IT
1.FIR filter has exact linear phase
2.FIR filter always stable
3.FIR filter can be realized in both recursive and non recursive structure
4.Filters wit h any arbitrary magnitude response can be tackled using FIR sequency
6. Define Phase Dealy April/May 2008 IT
When the input signal X(n) is applied which has non zero response
the output signal y(n) experience a delay with respect to the input
signal .Let the input signal be
X(n)=A , +
Where A= Maximum Amplitude of the signal
f=phase angle
Due to the delay in the system response ,the output signal lagging in phase but the
frequency remain the same
Y(n)= A ,
In This equation that the output is the time delayed signal and is more commonly known
as phase delayed at w=wo Is called phase delay

7. State the advantages and disadvantages of FIR filter over IIR filter.
(MAY 2006 IT DSP) & (NOV 2004
ECEDSP)
Advantages of FIR filter over IIR filter
- It is a stable filter
- It exhibit linear phase, hence can be easily designed.
- It can be realized with recursive and non-recursive structures
- It is free of limit cycle oscillations when implemented on a finite word length
digital system
Disadvantages of FIR filter over IIR filter
- Obtaining narrow transition band is more complex.
- Memory requirement is very high
- Execution time in processor implementation is very high.
8. List out the different forms of structural realization available for realizing a FIR system.
(MAY 2006 IT DSP)
The different types of structures for realization of FIR system are
1.Direct form-I 2. Direct form-II
9. What are the desirable and undesirable features of FIR Filters? (May/June 2006)-
ECE
The width of the main lobe should be small and it should contain as much of total
energy as possible.The side lobes should decease in energy rapidly as w tends to π
10. Define Hanning and Blackman window functions. (May/June 2006)-ECE
The window function of a causal hanning window is given by
W
Hann
(n) = 0.5 – 0.5cos2πn/ (M-1), 0≤n≤M-1
0, Otherwise
The window function of non-causal Hanning window I s expressed by
W
Hann
(n) = 0.5 + 0.5cos2πn/ (M-1), 0≤|n|≤(M-1)/2
0, Otherwise
The width of the main lobe is approximately 8π/M and thee peak of the first side lobe is
at -32dB.
The window function of a causal Blackman window is expressed by
W
B
(n) = 0.42 – 0.5 cos2πn/ (M-1) +0.08 cos4πn/(M-1), 0≤n≤M-1
= 0, otherwise
The window function of a non causal Blackman window is expressed by
W
B
(n) = 0.42 + 0.5 cos2πn/ (M-1) +0.08 cos4πn/(M-1), 0≤|n|≤(M-1)/2
= 0, otherwise
The width of the main lobe is approximately 12π/M and the peak of the first side lobe is
at -58dB.
11. What is the condition for linear phase of a digital filter? (APR 2005 ITDSP)
h(n) = h(M-1-n) Linear phase FIR filter with a nonzero response at ω=0
h(n) = -h(M-1-n)Low pass Linear phase FIR filter with a nonzero
response at ω=0
12. Define backward and forward predictions in FIR lattice filter. (NOV 2005 IT)
The reflection coefficient in the lattice predictor is the negative of the cross correlation
coefficients between forward and backward prediction errors in the lattice.
13. List the important characteristics of physically realizable filters. (NOV 2005 ITDSP)
Symmetric and anti- symmetric
- Linear phase frequency response
- Impulse invariance
14. Write the magnitude function of Butterworth filter. What is the effect of varying order of N
on magnitude and phase response? (Nov/Dec2005) -ECE
|H(jΏ)|
2
= 1 / [ 1 + (Ώ/Ώ
C
)
2N
] where N= 1,2,3,….

15. List the characteristics of FIR filters designed using window functions. NOV 2004
ITDSP

- the Fourier transform of the window function W(e
jw
) should have a small width
of main lobe containing as much of the total energy as possible
- the fourier transform of the window function W(e
jw
) should have side lobes that
decrease in energy rapidly as w to π. Some of the most frequently used window
functions are described in the following sections
16. Give the Kaiser Window function. (Apr/May 2004)-ECE
The Kaiser Window function is given by
W
K
(n) = I
0
(β) / I
0
(α) , for |n| ≤ (M-1)/2
Where α is an independent variable determined by Kaiser.
Β = α[ 1 – (2n/M-1)
2
]
17. What is meant by FIR filter? And why is it stable? (APR 2004 ITDSP)
FIR filter  Finite Impulse Response. The desired frequency response of a FIR
filter can be represented as

H
d
(e

)= Σ h
d
(n)e
-jωn

n= -∞
If h(n) is absolutely summable(i.e., Bounded Input Bounded Output Stable).
So, it is in stable.
18. Mention two transformations to digitize an analog filter. (APR 2004 ITDSP)
(i) Impulse-Invariant transformation techniques
(ii) Bilinear transformation techniques
19. Draw the direct form realization of FIR system. (NOV 2004
ITDSP)

20.Give the equation specifying Barlett and hamming window. (NOV 2004 ITDSP)
The transfer function of Barlett window
w
B
(n) = 1-(2|n|)/(N-1), ((N-1)/2)≥n≥-((N-1)/2)
The transfer function of Hamming window
w
hm
(n) = 0.54+0.46cos((2πn)/(N-1), ((N-1)/2)≥n≥-((N-1)/2) α = 0.54

UNIT-V - FINITE WORD LENGTH EFFECTS

1. Compare fixed point and floating point arithmetic. Nov/Dec 2008 CSE&MAY 2006 IT

2.What are the errors that arise due to truncation in floating point numbers
Nov/Dec 2008
CSE
1.Quantization error
2.Truncation error
Et=Nt-N
3.What are the effects of truncating an infinite flourier series into a finite series?
Nov/Dec 2008
CSE

4. Draw block diagram to convert a 500 m/s signal to 2500 m/s signal and state the problem
due to this conversion April/May2008
CSE

5.List errors due to finite world length in filter design April/May2008
CSE

- Input quantization error
- Product quantization error
- Coefficient quantization error

Fixed Point Arithmetic

Floating Point Arithmetic

- It covers only the dynamic
range.
- Compared to FPA, accuracy
is poor
- Compared to FPA it is low
cost and easy to design
- It is preferred for real time
operation system
- Errors occurs only for
multiplication

- Processing speed is high
- Overflow is rare
phenomenon
- It covers a large range of numbers
- It attains its higher accuracy
- Hardware implementation is costlier
and difficult to design
- It is not preferred for real time
operations.
- Truncation and rounding errors occur
- Processing speed is low
- Overflow is a range phenomenon
5. What do you mean by limit cycle oscillations in digital filter? Nov/Dec 2007
CSE
In recursive system the nonlinearities due to the finite precision arithmetic
operations often cause periodic oscillations to occur in the output ,even when the input
sequence is zero or some non zero constant value .such oscillation in recursive system
are called limit cycle oscillation
7.Define truncation error for sign magnitude representation and for 2’s complement
Representation April/May 2008 IT&APR 2005 IT
Truncation is a process of discarding all bits less significant than least significant bit
that is retained For truncation in floating point system the effect is seen only in
mantissa.if the mantissa is truncated to b bits ,then the error satisfies
0≥ > -2.2
-b
for x >0 and
0≤ < -2.2
-b
for x <0
8. What are the types of limit cycle oscillation? April/May 2008 IT
i.Zero input limit cycle oscillation
ii.overflow limit cycle oscillation
9. What is meant by overflow limit cycle oscillations? (May/Jun 2006 )
In fixed point addition, overflow occurs due to excess of results bit, which are
stored at the registers. Due to this overflow, oscillation will occur in the system. Thus
oscillation is called as an overflow limit cycle oscillation.
10. How will you avoid Limit cycle oscillations due to overflow in addition(MAY 2006 IT
DSP)
Condition to avoid the Limit cycle oscillations due to overflow in addition
|a
1
|+|a
2
|<1
a
1
and a
2
are the parameter for stable filter from stability triangle.
11.What are the different quantization methods? (Nov/Dec 2006)-ECE
- amplitude quantization
- vector quantization
- scalar quantization
12.List the advantages of floating point arithmetic. (Nov/Dec 2006)-ECE
- Large dynamic range
- Occurrence of overflow is very rare
- Higher accuracy
13.Give the expression for signal to quantization noise ratio and calculate the improvement
with an increase of 2 bits to the existing bit.
(Nov/Dec 2006, Nov/Dec 2005)-ECE
SNR
A / D
= 16.81+6.02b-20log
10
(R
FS

x
) dB.
With b = 2 bits increase, the signal to noise ratio will increase by 6.02
X 2 = 12dB.
14. What is truncation error? (APR 2005 ITDSP)
Truncation is an approximation scheme wherein the rounded number or
digits after the pre-defined decimal position are discarded.
15. What are decimators and interpolators? (APR 2005 ITDSP)
Decimation is a process of reducing the sampling rate by a factor D, i.e.,
down-sampling. Interpolation is a process of increasing the sampling rate by a
factor I, i.e., up-sampling.
16.What is the effect of down sampling on the spectrum of a signal?
(APR 2005 ITDSP) & (APR 2005 ITDSP)
The signal (n) with spectrum X(ω) is to be down sampled by the factor D. The
spectrum X(ω) is assumed to be non-zero in the frequency interval 0≤|ω|≤π.

17.Give the rounding errors for fixed and floating point arithmetic.
(APR 2004 ITDSP)
A number x represented by b bits which results in b
R
after being
Rounded off. The quantized error ε
R
due to rounding is given by
ε
R
=Q
R
(x)-x
where Q
R
(x) = quantized number(rounding error)
The rounding error is independent of the types of fixed point arithmetic, since
it involves the magnitude of the number. The rounding error is symmetric about
zero and falls in the range.
-((2
-bT
-2
-b
)/2)≤ ε
R
≤((2
-bT
-2
-b
)/2)
ε
R
may be +ve or –ve and depends on the value of x.
The error ε
R
incurred due to rounding off floating point number is in the range
-2
E
.2
-bR/2
)≤ ε
R
≤2
E
.2
-bR/2
18.Define the basic operations in multirate signal processing.
(APR 2004 ITDSP)
The basic operations in multirate signal processing are
(i)Decimation
(ii)Interpolation
Decimation is a process of reducing the sampling rate by a factor D, i.e., down-
sampling. Interpolation is a process of increasing the sampling rate by a factor I,
i.e., up-sampling.

19. Define sub band coding of speech. (APR 2004 ITDSP)
& (NOV 2003 ECEDSP) & (NOV 2005 ECEDSP)
Sub band coding of speech is a method by which the speech signal is
subdivided into several frequency bands and each band is digitally encode
separately. In the case of speech signal processing, most of its energy is
contained in the low frequencies and hence can be coded with more bits then
high frequencies.

20.What is the effect of quantization on pole locations? (NOV 2004 ITDSP)
N
D(z) = Π (1-p
k
z
-1
)
k=1
▲pk is the error or perturbation resulting from the quantization of the filter
coefficients

21.What is an anti-imaging filter? (NOV 2004 ITDSP)
The image signal is due to the aliasing effect. In caseof decimation by M,
there will be M-1 additional images of the input spectrum. Thus, the input
spectrum X(ω) is band limited to the low pass frequency response. An anti-
aliasing filter eliminates the spectrum of X(ω) in the range (л/D≤ ω ≤π.
The anti-aliasing filter is LPF whose frequency response H
LPF
(ω) is given by
H
LPF
(ω) = 1, |ω|≤ л/M
= 0, otherwise.
D  Decimator
22.What is a decimator? If the input to the decimator is x(n)={1,2,-1,4,0,5,3,2}, What is the
output? (NOV 2004 ITDSP)
Decimation is a process of reducing the sampling rate by a factor D, I.e., down-
sampling.
x(n)={1,2,-1,4,0,5,3,2}
D=2
Output y(n) = {1,-1,0,3}
23.What is dead band? (Nov/Dec 2004)-ECE
In a limit cycle the amplitude of the output are confined to a range of value,
24.How can overflow limit cycles be eliminated? (Nov/Dec 2004)-ECE
- Saturation Arithmetic
- Scaling
25.What is meant by finite word length effects in digital filters?
(Nov/Dec 2003)-ECE
The digital implementation of the filter has finite accuracy. When numbers are
represented in digital form, errors are introduced due to their finite accuracy. These
errors generate finite precision effects or finite word length effects.
When multiplication or addition is performed in digital filter, the result is to be
represented by finite word length (bits). Therefore the result is quantized so that it can
be represented by finite word register. This quantization error can create noise or
oscillations in the output. These effects are called finite word length effects.

PART B
UNIT-1 - SIGNALS AND SYSTEMS

1.Determine whether the following signals are Linear ,Time Variant, causal and stable
(1) Y(n)=cos[x(n)] Nov/Dec 2008 CSE
(2) Y(n)=x(-n+2)
(3) Y(n)=x(2n)
(4) Y(n)=x(n)+nx(n+1)
Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.79)
2. Determine the causal signal x(n) having the Z transform Nov/Dec 2008 CSE
X(z)=
Refer book : Digital signal processing by Ramesh Babu .(Pg no 2.66)
3. Use convolution to find x(n) if X(z) is given by Nov/Dec 2008 CSE
for ROC
Refer book : Digital signal processing by Ramesh Babu .(Pg no 2.62)

4.Find the response of the system if the input is {1,4,6,2} and impulse response of the
system is {1,2,3,1}
April/May2008CSE

Refer book: Digital signal processing by A.Nagoor kani .(Pg no 23-24)

5.find r
xy
and r
yx
for x={1,0,2,3} and y={4,0,1,2}. April/May2008
CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.79)

6. (i) Check whether the system y(n)=ay(n-1)+x(n) is linear ,casual,
shift variant, and stable
Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.51-1.57)

(ii) Find convolution of {5,4,3,2} and {1,0,3,2} April/May2008 CSE

Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.79)

7. (i) Compute the convolution y(n) of the signals
x(n)= a
n
, -3≤n≤5
0 , elsewhere

and

h (n)= 1, 0≤n≤4
0, elsewhere Nov/Dec 2007 CSE

8.A discrete-time system can be static or dynamic, linear or nonlinear,
Time invariant or time varying, causal or non causal, stable or unstable. Examine
the following system with respect to the properties also.
(1) y(n) = cos [x(n)]
(2) y(n)=x(-n+2)
(3) y(n)=x(2n)
(4) y(n)=x(n).cosWo(n)
Nov/Dec 2007 CSE

Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.185-1.197)

9.(i) Determine the response of the casual system.
y(n)-y(n-1)=x(n)+x(n-1) to inputs x(n)=u(n) and x(n)=2-n u(n). Test its
stability.
(ii) Determine the IZT of X(z)=1 / [(1-z-1)(1-z-1)2] Nov/Dec 2007 CSE

Refer book : Digital signal processing by A.Nagoor kani .
(Pg no 463)
10.(i)Determine the Z-transform of the signal x(n)=a
n
u(n)-b
n
u(-n-1), b>a and plot the
ROC.
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number (157)

(ii) Find the steady state value given Y(z)={0.5/[(1-0.75z
-1
)(1-z
-1
)]}
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number (207)
(iii) Find the system function of the system described by
y(n)-0.75y(n-1)+0.125y(n-2)=x(n)-x(n-1) and plot the poles and zeroes of

11.(i) find the convolution and correlation for x(n)={0,1,-2,3,-4} and h(n)={0.5,1,2,1,0.5}.

Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.79)
(ii)Determine the Impulse response for the difference equation
Y(n) + 3 y(n-1)+2y(n-2)=2x(n)-x(n-1) April/May2008 IT
Refer book : Digital signal processing by Ramesh Babu .(Pg no 2.57)
12. (i) Compute the z-transform and hence determine ROC of x(n) where

X (n) = (1/3)
n
u(n).n ≥ 0

(1/2)
-n
u(n).n<0

Refer book : Digital signal processing by Ramesh Babu .(Pg no 2.20)
(iii) prove the property that convolution in Z-domains multiplication in time domain
April/May2008 IT

Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.77)
13.Find the response of the system if the input is {1,4,6,2} and impulse response of the
system is {1,2,3,1} April/May2008CSE

Refer book: Digital signal processing by A.Nagoor kani .(Pg no 23-24)

14.find r
xy
and r
yx
for x={1,0,2,3} and y={4,0,1,2}. April/May2008 CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.79)
15.(i) Check whether the system y(n)=ay(n-1)+x(n) is linear ,casual,
shift variant, and stable
Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.51-1.57)

(ii) Find convolution of {5,4,3,2} and {1,0,3,2} April/May2008 CSE

Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.79)

16. (i) Compute the convolution y(n) of the signals
x(n)= a
n
, -3≤n≤5
0 , elsewhere

and

h (n)= 1, 0≤n≤4
0, elsewhere Nov/Dec 2007 CSE

17.A discrete-time system can be static or dynamic, linear or nonlinear,
Time invariant or time varying, causal or non causal, stable or unstable. Examine
the following system with respect to the properties also.
(1) y(n) = cos [x(n)]
(2) y(n)=x(-n+2)
(3) y(n)=x(2n)
(4) y(n)=x(n).cosWo(n)
Nov/Dec 2007 CSE

Refer book : Digital signal processing by Ramesh Babu .(Pg no 1.185-1.197)
18.(i) Determine the response of the casual system.
y(n)-y(n-1)=x(n)+x(n-1) to inputs x(n)=u(n) and x(n)=2-n u(n). Test its
stability.
(ii) Determine the IZT of X(z)=1 / [(1-z-1)(1-z-1)2] Nov/Dec 2007 CSE

Refer book : Digital signal processing by A.Nagoor kani .
(Pg no 463)
19.(i)Determine the Z-transform of the signal x(n)=a
n
u(n)-b
n
u(-n-1), b>a and plot the
ROC.
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number (157)

(ii) Find the steady state value given Y(z)={0.5/[(1-0.75z
-1
)(1-z
-1
)]}
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number (207)
(iii) Find the system function of the system described by
y(n)-0.75y(n-1)+0.125y(n-2)=x(n)-x(n-1) and plot the poles and zeroes of
H(z). (MAY 2006 ITDSP)
Refer signals and systems by P. Ramesh babu , page no:10.65
(To find the impulse response h(n) and take z-transform.)

20.(i)Using Z-transform, compute the response of the system
y(n)=0.7y(n-1)-0.12y(n-2+x9n-1)+x(n-2) to the input x(n)=nu(n). Is the system
stable?
Refer signals and systems by chitode, page no:4.99
(ii)State and prove the properties of convolution sum. (MAY 2006 ECESS)
Refer signals and systems by chitode, page no:4.43 to 4.45

21.State and prove the sampling theorem. Also explain how reconstruction of original signal
is done from the sampled signal. (NOV 2006 ECESS)
Refer signals and systems by chitode, page no:3-2 to 3-7

22.Explain the properties of an LTI system. (NOV 2006 ECESS)
Refer signals and systems by chitode, page no:4.47 to 4.49

23.a. Find the convolution sum for the x(n) =(1/3)
-n
u(-n-1) and h(n)=u(n-1)
Refer signals and systems by P. Ramesh babu , page no:3.76,3.77
b. Convolve the following two sequences linearly x(n) and h(n) to get y(n).
x(n)= {1,1,1,1} and h(n) ={2,2}.Also give the illustration
Refer signals and systems by chitode, page no:67
c. Explain the properties of convolution. (NOV2006 ECESS)
Refer signals and systems by chitode, page no:4.43 to 4.45

24. Check whether the following systems are linear or not
1. y(n) = x
2
(n)
2. y(n) = nx(n) (APRIL 2005 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition.
Page number (67)
25.(i)Determine the response of the system described by,
y(n)-3y(n-1)-4y(n- 2)=x(n)+2x(n-1) when the input sequence is x(n)=4
n
u(n).
Refer signals and systems by P. Ramesh babu , page no:3.23
(ii)Write the importance of ROC in Z transform and state the relationship between Z
transforms to Fourier transform. (APRIL 2004 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition.
Page number (153)
Refer S Poornachandra & B Sasikala, “Digital Signal Processing”,
Page number (6.10)

UNIT-II - FAST FOURIER TRANSFORMS

1.By means of DFT and IDFT ,Determine the sequence x3(n) corresponding to the circular
convolution of the sequence x1(n)={2,1,2,1}.x2(n)={1,2,3,4}. Nov/Dec 2008 CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no 3.46)
2. State the difference between overlap save method and overlap Add method
Nov/Dec 2008 CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no 3.88)
3. Derive the key equation of radix 2 DIF FFT algorithm and draw the relevant flow graph
taking the computation of an 8 point DFT for your illustration Nov/Dec 2008
CSE
Refer book : Digital signal processing by Nagoor Kani .(Pg no 215)
4. Compute the FFT of the sequence x(n)=n+1 where N=8 using the in place radix 2
decimation in frequency algorithm. Nov/Dec 2008 CSE
Refer book : Digital signal processing by Nagoor Kani .(Pg no 226)
5. Find DFT for {1,1,2,0,1,2,0,1} using FFT DIT butterfly algorithm and
plot the spectrum April/May2008
CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no 4.17)
6. (i)Find IDFT for {1,4,3,1} using FFT-DIF method April/May2008
CSE
(ii)Find DFT for {1,2,3,4,1} (MAY 2006
ITDSP)
Refer book : Digital signal processing by Ramesh Babu .(Pg no 4.29)
7.Compute the eight point DFT of the sequence x(n)={ ½,½,½,½,0,0,0,0} using radix2
decimation in time and radix2 decimation in frequency algorithm. Follow exactly the
corresponding signal flow graph and keep track of all the intermediate quantities by
putting them on the diagram. Nov/Dec 2007 CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no 4.30)
8.(i) Discuss the properties of DFT.
Refer book : Digital signal processing by S.Poornachandra.,B.sasikala.
(Pg no 749)
(ii)Discuss the use of FFT algorithm in linear filtering. Nov/Dec 2007
CSE
Refer book : Digital signal processing by John G.Proakis .(Pg no 447)

9.(i) if x(n) N pt DFT X(k) then, prove that

X1(n)x2(n)=1/N [Xt(k) X2(k)]. April/May2008 IT

Refer book : Digital signal processing by Ramesh Babu .(Pg no 3.34)
(ii) Find 8 Point DFT of x(n)=0.5,0≤n≤3 Using DIT FFT
0, 4≤n≤7 April/May2008 IT
Refer book : Digital signal processing by Ramesh Babu .(Pg no 4.32)
10.Derive the equation for radix 4 FFT for N=4 and Draw the butterfly Diagram.
April/May2008 IT
11. (i) Compute the 8 pt DFT of the sequence
Refer P. Ramesh babu, “Signals and Systems”.Page number (8.89)
(ii) Determine the number of complex multiplication and additions involved in a N-
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing Principles,
Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition. Page number
(456 & 465)
12.Find the 8-pt DFT of the sequence x(n)={1,1,0,0} (APRIL 2005
ITDSP)
Refer P. Ramesh babu, “Signals and Systems”. Page number (8.58)
13.Find the 8-pt DFT of the sequence
x(n)= 1, 0≤n≤7
0, otherwise
using Decimation-in-time FFT algorithm (APRIL 2005 ITDSP)
Refer P. Ramesh babu, “Signals and Systems”.Page number (8.87)
14.Compute the 8 pt DFT of the sequence
x(n)={0.5,0.5,0.5,0.5,0,0,0,0} using DIT FFT (NOV 2005 ITDSP)
Refer P. Ramesh babu, “Signals and Systems”.Page number (8.89)
N
15.By means of DFT and IDFT , determine the response of an FIR filter with impulse
response h(n)={1,2,3},n=0,1,2 to the input sequence x(n) ={1,2,2,1}.
(NOV 2005 ITDSP)
Refer P. Ramesh babu, “Signals and Systems”.Page number (8.87)
16.(i)Determine the 8 point DFT of the sequence
x(n)= {0,0,1,1,1,0,0,0}
Refer P. Ramesh babu, “Signals and Systems”.Page number (8.58)
(ii)Find the output sequence y(n) if h(n)={1,1,1} and x(n)={1,2,3,4} using circular
convolution (APR 2004 ITDSP)
Refer P. Ramesh babu, “Signals and Systems”.Page number (8.65)
17. (i)What is decimation in frequency algorithm? Write the similarities and differences
between DIT and DIF algorithms. (APR 2004 ITDSP) & (MAY 2006 ECEDSP)
Refer P. Ramesh babu, “Signals and Systems”. Page number (8.70-8.80)
18.Determine 8 pt DFT of x (n)=1for -3≤n≤3 using DIT-FFT algorithm (APR 2004
ITDSP)
Refer P. Ramesh babu, “Signals and Systems”. Page number (8.58)
19.Let X(k) denote the N-point DFT of an N-point sequence x(n).If the DFT of X(k)is
computed to obtain a sequence x
1
(n). Determine x
1
(n) in terms of x(n) (NOV 2004
ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing Principles,
Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition. Page number (456 &
465)

UNIT-III - IIR FILTER DESIGN

1.Design a digital filter corresponding to an analog filter H(s)= using the impulse
invariant method to work at a sampling frequency of 100 samples/sec
Nov/Dec 2008 CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no5.40)
2.Determine the direct form I ,direct form II ,Cascade and parallel structure for the system
Y(n)=-0.1y(n-1)+0.72y(n-2)+0.7x(n)-0.25x(n-2) Nov/Dec 2008 CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no5.61)
3.What is the main drawback of impulse invariant method ?how is this overcome by
bilinear transformation? Nov/Dec 2008 CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no5.46)
4.Design a digital butter worth filter satisfying the constraints Nov/Dec 2008 CSE

0.707≤ ≤1 for 0 ≤w≤
≤0.20 for ≤w≤
With T=1 sec using bilinear transformation .realize the same in Direct form II

Refer book : Digital signal processing by Ramesh Babu .(Pg no5.79)
5. (i)Design digital filter with H(s) = using T=1sec.
(ii)Design a digital filter using bilinear transform for H(s)=2/(s+1)(s+2)with cutoff
frequency as 100 rad/sec and sampling time =1.2 ms April/May2008
CSE
Refer book : Digital signal processing by A.Nagoor kani .(Pg no 341)
6. (i) Realize the following filter using cascade and parallel form with
direct form –I structure

1+z
-1
+z
-2+
5z
-3

( 1+Z
-1
)(1+2Z
-1
+4Z
-2
)

( ii) Find H(s) for a 3
rd
order low pass butter worth filter April/May2008
CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no 5.8)
7.(i) Derive bilinear transformation for an analog filter with system function
H(s) =b / (s+a)
Refer book: Digital signal processing by John G.Proakis .(Pg no 676-679)
(ii)Design a single pole low pass digital IIR filter with -3 db bandwidth of
0.2п by use of bilinear transformation. Nov/Dec 2007
CSE

8.(i) Obtain the Direct Form I, Direct Form II, cascade and parallel realization for the
following system Y(n)= -0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2)
Refer book : Digital signal processing by Ramesh Babu .(Pg no 5.68)
(ii) Discuss the limitation of designing an IIR filter using impulse
invariant method. Nov/Dec 2007 CSE

Refer book : Digital signal processing by A.Nagoor kani . (Pg no 330)
9. Design a low pass Butterworth filter that has a 3 dB cut off frequency of 1.5 KHz and an
attenuation of 40 dB at 3.0 kHz April/May2008 IT
Refer book : Digital signal processing by Ramesh Babu .(Pg no 5.14)
10. (i) Use the Impulse invariance method to design a digital filter from an analog
prototype that has a system function
April/May2008 IT
Ha(s)=s+a/((s+a)
2
+b
2
)
Refer book : Digital signal processing by Ramesh Babu .(Pg no 5.42)
(ii) Determine the order of Cheybshev filter that meets the following specifications
(1) 1 dB ripple in the pass band 0≤|w| ≤ 0.3 b
(2) Atleast 60 dB attrnuation in the stop band 0.35∏ ≤|w| ≤∏ Use Bilinear
Transformation
Refer book : Digital signal processing by Ramesh Babu .(Pg no 5.27)
11.(i) Convert the analog filter system functionH
a
(s)={(s+0.1)/[(s+0.1)
2
+9]} into a digital IIR
filter using impulse invariance method.(Assume T=0.1sec) (APR 2006 ECEDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing Principles,
Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition. Page number
(675)
12.Determine the Direct form II realization for the following system:
y(n)=-0.1y(n-1+0.72y(n-2)+0.7x(n)-0.252x(n-2). (APRIL 2005 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing Principles,
Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition. Page number
(601-7.9b)

13.Explain the method of design of IIR filters using bilinear transform method.
(APRIL 2005 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition. Page
number (676-8.3.3)
14.Explain the following terms briefly:
(i)Frequency sampling structures
(ii)Lattice structure for IIR filter (NOV 2005 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition. Page
number (506 &531)
15.Consider the system described by
y(n)-0.75y(n-1)+0.125y(n-2)=x(n)+0.33x(n-1).
Determine its system function (NOV 2005 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition. Page
number (601-7.37)
16.Find the output of an LTI system if the input is x(n)=(n+2) for 0≤n≤3 and h(n)=a
n
u(n) for
all n (APR 2004 ITDSP)
Refer signals and systems by P. Ramesh babu , page no:3.38
17.Obtain cascade form structure of the following system:
y(x)=-0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2) (APR 2004 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition.
Page number (601-7.9c)
18.Verify the Stability and causality of a system with
H(z)=(3-4Z
-1
)/(1+3.5Z
-1
+1.5Z
-2
) (APR 2004 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition.
Page number (209)

UNIT-IV - FIR FILTER DESIGN

1.Design a FIR linear phase digital filter approximating the ideal frequency response
Nov/Dec 2008 CSE

With T=1 Sec using bilinear transformation .Realize the same in Direct form II
Refer book : Digital signal processing by Nagoor Kani .(Pg no 367)
2.Obtain direct form and cascade form realizations for the transfer function of the system
given by

Nov/Dec 2008
CSE
Refer book : Digital signal processing by Nagoor Kani .(Pg no 78)
3.Explain the type I frequency sampling method of designing an FIR filter.
Nov/Dec 2008
CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no6.82)

4.Compare the frequency domain characteristics of various window functions .Explain how
a linear phase FIR filter can be used using window method. Nov/Dec 2008 CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no6.28)

5. Design a LPF for the following response .using hamming window with
N=7
April/May2008 CSE

6. (i) Prove that an FIR filter has linear phase if the unit sample response satisfies the
condition h(n)= ±h(M-1-n), n=0,1,….M-1. Also discuss symmetric and antisymmetric cases
of FIR filter. Nov/Dec 2007

Refer book: Digital signal processing by John G.Proakis .
(Pg no 630-632)
(ii) Explain the need for the use of window sequences in the design of FIR filter. Describe
the window sequences generally used and compare their properties.
Nov/Dec 2007 CSE
Refer book : Digital signal processing by A.Nagoor kani .(Pg no 292-295)
7.(I) Explain the type 1 design of FIR filter using frequency sampling
technique. Nov/Dec 2007 CSE
Refer book : Digital signal processing by A.Nagoor kani .(Pg no 630-632)
(ii)A low pass filter has the desired response as given below

e
-i3w
, 0≤w<∏/2
Hd(e
jw
)=
0, ∏/2≤<∏
Determine the filter coefficients h(n) for M=7 using frequency sampling
method.
Nov/Dec 2007
CSE
8.(i) For FIR linear phase Digital filter approximating the ideal frequency response

Hd(w) = 1 ≤|w| ≤∏ /6

0 ∏ /6≤ |w| ≤∏
Determine the coefficients of a 5 tap filter using rectangular Window
Refer book : Digital signal processing by A.Nagoor kani .(Pg no 415
(ii) Determine the unit sample response h(n) of a linear phase FIR filter of Length M=4
for which the frequency response at w=0 and w= ∏/2 is given as Hr(0) ,Hr(∏/2) =1/2
April/May2008 IT
Refer book : Digital signal processing by A.Nagoor kani .(Pg no 310)
9.(i) Determine the coefficient h(n) of a linear phase FIR filter of length M=5 which has
symmetric unit sample response and frequency response
Hr(k)=1 for k=0,1,2,3
0.4 for k=4
0 for k=5, 6, 7 April/May2008 IT(NOV 2005 ITDSP)
Refer book : Digital signal processing by A.Nagoor kani .(Pg no 308)

m-1
(ii) Show that the equation ∑ h(n)=sin (wj-wn)=0,is satisfied for a linear phase FIR filter

n=0
of length 9
April/May2008 IT
10. Design linear HPF using Hanning Window with N=9

H(w) =1 -п to Wc and Wc to п
= 0 otherwise
April/May2008 IT
Refer book : Digital signal processing by A.Nagoor kani .(Pg no 301)
11.Explain in detail about frequency sampling method of designing an FIR filter.
(NOV 2004 ITDSP) & ( NOV 2005 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number (630)
12.Explain the steps involved in the design of FIR Linear phase filter using window method.
(APR 2005 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number (8.2.2 & 8.2.3)
13.(i)What are the issues in designing FIR filter using window method?
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number (8.2)
(ii)An FIR filter is given by
y(n)=2x(n)+(4/5)x(n-1)+(3/2)x(n-2)+(2/3)x(n-3) find the lattice structure
coefficients (APR 2004 ITDSP)
Refer S Poornachandra & B Sasikala, “Digital Signal Processing”,
Page number (FIR-118)

UNIT-V - FINITE WORD LENGTH EFFECTS
1.Draw the circuit diagram of sample and hold circuit and explain its operation
Nov/Dec 2008 CSE/ Nov/Dec 2007
CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no1.172)
2. The input of the system y(n)=0.99y(n-1)+x(n) is applied to an ADC .what is the power
produced by the quantization noise at the output of the filter if the input is quantized to 8
bits
Nov/Dec 2008
CSE
Refer book : Digital signal processing by Nagoor Kani .(Pg no 423)
3.Discuss the limit cycle in Digital filters Nov/Dec 2008 CSE
Refer book : Digital signal processing by Nagoor Kani .(Pg no 420)
4.What is vocoder? Explain with a block diagram Nov/Dec 2008 CSE/ Nov/Dec 2007 CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no10.7)

(ii) Discuss about multirate Signal processing April/May 2008 CSE
Refer book : Digital signal processing by Ramesh Babu .(Pg no 8.1)
5. (i) Explain how the speech compression is achieved .
(ii) Discuss about quantization noise and derive the equation for
finding quantization noise power. April/May2008CSE
Refer book : Digital signal processing by Ramesh Babu.(Pg no 7.9-7.14)
6. Two first order low pass filter whose system functions are given below are connected in
cascade. Determine the overall output noise
power. H1(z) = 1/ (1-0.9z-1) and H2(z) = 1/ (1-0.8z-1) Nov/Dec 2007 CSE
Refer book: Digital signal processing by Ramesh Babu. (Pg no 7.24)
7. Describe the quantization errors that occur in rounding and
truncation in two’s complement. Nov/Dec 2007 CSE
Refer book : Digital signal processing by John G.Proakis .(Pg no 564)
m
8. Explain product quantization and prove б
err
2
=∑

б
2
oi
April/May2008 IT
i=1
Refer book : Digital signal processing by A.Nagoor kani .(Pg no 412)
9.A cascade Realization of the first order digital filter is shown below ,the system function
of the individual section are H1(z)=1/(1-0.9z
-1
) and H2(z) =1/(1-0.8z
-1
) .Draw the product
quantization noise model of the system and determine the overall output noise power

April/May2008 IT
Refer book : Digital signal processing by A.Nagoor kani .(Pg no 415)
9. (i) Show dead band effect on y(n) = .95 y(n-1)+x(n) system restricted to 4 bits .Assume
x(0) =0.75 and y(-1)=0
Refer book : Digital signal processing by A.Nagoor kani .(Pg no 423-426)
11. Explain the following terms briefly:
(i)Perturbation error
(ii)Limit cycles (NOV 2005 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition.
Page number(7.7.1 &7.7.2)
12.(i) Explain clearly the downsampling and up sampling in multirate signal
processing. (APRIL 2005 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number (784-790)
(ii)Explain subband coding of speech signal
(NOV 2003 ITDSP) & (NOV 2004 ITDSP) & (NOV 2005 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number(831-833)
13.(i) Derive the spectrum of the output signal for a decimator
(ii) Find and sketch a two fold expanded signal y(n) for the input
(APR 2004 ITDSP) &(NOV 2004 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number (788)
14.(i)Propose a scheme for sampling rate conversion by a rational factor I/D.
(NOV 2004 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd

Edition. Page number (790)
15. Write applications of multirate signal processing in Musical sound processing
(NOV 2004 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition.
Page number (952)
16. With examples illustrate (i) Fixed point addition (ii) Floating point multiplication (iii)
Truncation (iv) Rounding.(APR 2005 ITDSP) & (NOV 2003 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition.
Page number (7.5)
17. Describe a single echo filter using in musical sound processing.
(APRIL 2004 ITDSP)
Refer John G Proakis and Dimtris G Manolakis, “Digital Signal Processing
Principles, Algorithms and Application”, PHI/Pearson Education, 2000, 3
rd
Edition.
Page number (12.5.3)

STAFF INCHARGE HOD/ECE

27. What is an anti-aliasing filter? 28. What is the necessary and sufficient condition on the impulse response for stability? (EC 333, May „07) 22. State the condition for a digital filter to be causal and stable

Part-B 1. a) Compute the convolution y(n) of the signals x(n)= an -3≤n≤5 0 elsewhere and h(n)= 1 0≤n≤4 0 elsewhere (AU DEC 07)

b) A discrete time system can be static or dynamic, linear or non-linear, Time variant or time invariant, causal or non causal, stable or unstable. Examine the following system with respect to the properties also (AU DEC 07) 1) y(n)=cos(x(n)) 2) y(n)=x(-n+2) 3) y(n)=x(2n) 4)y(n)=x(n) cos ωn 2. a) Determine the response of the causal system y(n)-y(n-1)=x(n)+x(n-1) to inputs x(n)=u(n) and x(n)=2 –n u(n).Test its stability b) Determine the IZT of X(Z)=1/(1-Z-1)(1-Z-1)2 (AU DEC 07) Determine whether each of the following systems defined below is (i) casual (ii) linear (iii) dynamic (iv) time invariant (v) stable (a) y(n) = log10[{x(n)}] (b) y(n) = x(-n-2) (c) y(n) = cosh[nx(n) + x(n-1)]

3. Compute the convolution of the following signals x(n) = {1,0,2,5,4} h(n) = {1,-1,1,-1} ↑ ↑ h(n) = {1,0,1} x(n) = {1,-2,-2,3,4} ↑ ↑ 4. Find the convolution of the two signals x(n) = 3nu(-n); h(n) = (1/3)n u(n-2) x(n) = (1/3) –n u(-n-1); h(n) = u(n-1) x(n) = u(n) –u(n-5); h(n) = 2[u(n) – u(n-3)]

5. Find the discrete-time Fourier transform of the following x(n) = 2-2n for all n x(n) = 2nu(-n) x(n) = n [1/2] (n)

6. Determine and sketch the magnitude and phase response of the following systems (a) y(n) = 1/3 [x(n) + x(n-1) + x(n-2)] (b) y(n) = ½[x(n) – x(n-1)] (c) y(n) - 1/2y(n-1)=x(n) 7. a) Determine the impulse response of the filter defined by y(n)=x(n)+by(n-1) b) A system has unit sample response h(n) given by h(n)=-1/δ(n+1)+1/2δ(n)-1-1/4 δ(n-1). Is the system BIBO stable? Is the filter causal? Justify your answer (DEC 2003) 8. Determine the Fourier transform of the following two signals(CS 331 DEC 2003) a) a n u(n) for a<1 b) cos ωn u(n) 9. Check whether the following systems are linear or not a) y(n)=x 2 (n) b) y(n)=n x(n) (AU APR 05)

10. For each impulse response listed below, dtermine if the corresponding system is i) causal ii) stable (AU MAY 07) 1) 2 n u(-n) 2) sin nЛ/2 (AU DEC 04) 3) δ(n)+sin nЛ 4) e 2n u(n-1) 11. Explain with suitable block diagram in detail about the analog to digital conversion and to reconstruct the analog signal (AU DEC 07) 12. Find the cross correlation of two sequences x(n)={1,2,1,1} y(n)={1,1,2,1}

(AU DEC 04)

13. Determine whether the following systems are linear , time invariant 1) y(n)=A x(n)+B 2) y(n)=x(2n) Find the convolution of the following sequences: (AU DEC 04) 1) x(n)=u(n) h(n)=u(n-3) 2) x(n)={1,2,-1,1} h(n)={1,0,1,1}

UNIT II FAST FOURIER TRANSFORMS
1) THE DISCRETE FOURIER TRANSFORM

PART A 1. Find the N-point DFT of a sequence x(n) ={1 ,1, 2, 2} 2. Determine the circular convolution of the sequence x1(n)={1,2,3,1} and x2(n)={4,3,2,1} (AU DEC 07) 3. Draw the basic butterfly diagram for radix 2 DIT-FFT and DIF-FFT(AU DEC 07) 4. Determine the DTFT of the sequence x(n)=a n u(n) for a<1 (AU DEC 06) 5. Is the DFT of the finite length sequence periodic? If so state the reason (AU DEC 05) 6. Find the N-point IDFT of a sequence X(k) ={1 ,0 ,0 ,0} (Oct 98) 7. what do you mean by „in place‟ computation of FFT? (AU DEC 05) 8. What is zero padding? What are its uses? (AU DEC 04) 9. List out the properties of DFT (MU Oct 95,98,Apr 2000) 10. Compute the DFT of x(n)=∂(n-n0) 11. Find the DFT of the sequence of x(n)= cos (n∏/4) for 0≤n≥ 3 (MU Oct 98) 12. Compute the DFT of the sequence whose values for one period is given by x(n)={1,1,-2,-2}. (AU Nov 06,MU Apr 99) 13. Find the IDFT of Y(k)={1,0,1,0} (MU Oct 98) 14. What is zero padding? What are its uses? 15. Define discrete Fourier series. 16. Define circular convolution 17. Distinguish between linear convolution and Circular Convolution. (MU Oct 96,Oct 97,Oct 98) 18. Obtain the circular convolution of the following sequences x(n)={1, 2, 1} and h(n)={1, -2, 2} 19. Distinguish between DFT and DTFT (AU APR 04) 20. Write the analysis and synthesis equation of DFT (AU DEC 03) 21. Assume two finite duration sequences x1(n) and x2(n) are linearly combined. What is the DFT of x3(n)?(x3(n)=Ax1(n)+Bx2(n)) (MU Oct 95) 22. If X(k) is a DFT of a sequence x(n) then what is the DFT of real part of x(n)? 23. Calculate the DFT of a sequence x(n)=(1/4)^n for N=16 (MU Oct 97) 24. State and prove time shifting property of DFT (MU Oct 98) 25. Establish the relation between DFT and Z transform (MU Oct 98,Apr 99,Oct 00) 26. What do you understand by Periodic convolution? (MU Oct 00) 27. How the circular convolution is obtained using concentric circle method? (MU Apr 98) 28. State the circular time shifting and circular frequency shifting properties of DFT 29. State and prove Parseval‟s theorem 30. Find the circular convolution of the two sequences using matrix method X1(n)={1, 2, 3, 4} and x2(n)={1, 1, 1, 1}

0. x2(n)={1.1} using circular convolution (AU APR 04) .1. 1.3. 2 . Determine IDFT of the following (a)X(k)={1.1-j2. 1} 35.1.1.2}.0.1+j2} (b)X(k)={1. 1.-2-j. Find 8-point DFT of the following sequences (a) x(n)={1.2.2}. (c) x1(n)=sin n∏/2.-1.3.2.1.1.-1.-2} Determine the response of the LTI system by radix2 DIT-FFT? If the impulse response of a LTI system is h(n)=(1.-2.h2(n)=(1/4)^n*u(n) (AU May 07) 7. 0.0.0. 6.1. Find 4-point DFT of the following sequences (a) x(n)={1. 3. (b) x1(n)={2. State the time reversal property of DFT 32.-2+j} (AU DEC 06) 4.2. Find the output sequence y(n)if h(n)={1.-2} (c) x(n)=2n (d) x(n)=sin(n∏/2) 2.2. x2(n)={-1. State circular convolution and circular correlation properties of DFT 34. The first five coefficients of X(K)={1.1.-1.1}.2} 3.2+5j.0} (b) x(n)={1. x2(n)=3n 0≤n≥7 5.1.0. 2.3.1.0} (b) x(n)={1.-1. 1.-1) (AU Nov 06). 8.-1. Find the circular convolution of the following using matrix method and concentric circle method (a) x1(n)={1. If the DFT of x(n) is X(k) then what is the DFT of x*(n)? 33.31.2.0} (c)X(k)={1. 1. Find the circular convolution of the following two sequences using concentric circle method x1(n)={1. 4} and x2(n)={1. 1} and prove that it is equal to the linear convolution of the same. Determine the impulse response for the cascade of two LTI systems having impulse responses h1(n)=(1/2)^n* u(n). 2.3}.1} and x(n)={1.Calculate the DFT of the sequence x(n)={1.-2. 3. Determine the circular convolution of the two sequences x1(n)={1.5 }Find the remaining coefficients PART B 1. 2+3j.1.0. 4} x2(n)={1.

9y(n-1) (AU APR 05) 14.9.0.0.2. Find the DFT of sequence x(n)={1.1.2.0.0.1. 10.1.2.1.5. 1} using DIF-FFT algorithms (AU DEC 04) PART B 1.1.0. Why FFT is needed? (AU DEC 03) (MU Oct 95. 4.3.1} (AU APR 05) (c)x(n)={0.2} (e)x(n)={0.0} using DIT-FFT algorithms 19.3. 98). Draw the flow graph of a two point DFT for a DIT decomposition.1.1. Find the DFT of sequence x(n)={1. 1. What is decimation-in-time algorithm? (MU Oct 95).5x(n)+0.0.0. (MU Oct 97.1. 3.1. Draw the direct form realization of FIR system (AU DEC 04) 9.0} (AU APR 04) . How many multiplications and additions are required to compute N-point DFT using radix 2 FFT? (AU DEC 04) 8.3.1.5.2. What are the applications of FFT algorithms? 18.5.1} 2) FAST FOURIER TRANSFORM PART A 1.0} (b)x(n)={1. Draw the basic butterfly diagram for DIT and DIF algorithm.3. What is the main advantage of FFT? 6. 12.2.0. What is decimation-in-frequency algorithm? (MU Oct 95. Draw the direct form realization of the system y(n)=0. (AU 07).4} x2(n)={4. 16.0. State and prove the following properties of DFT (AU DEC 03) 1) Cirular convolution 2) Parseval‟s relation 2) Find the circular convolution of x1(n)={1.3. 15.Apr 98).-1.2. Compute an 8-point DFT of the following sequences using DIT and DIF algorithms (a)x(n)={1. How do we can calculate IDFT using FFT algorithm? 17.Apr 98) What is FFT? (AU DEC 06) Obtain the block diagram representation of the FIR filter (AU DEC 06) Calculate the number of multiplications needed in the calculation of DFT and FFT with 64 point sequence.0. What is FFT? (AU Nov 06) 7. What do you mean by „in place‟ computation in DIT-FFT algorithm? (AU APR 04) 11.5.0. 5.1.1. Mention the advantage of direct and cascade structures (AU APR 04) 13. 2.-1.1.0} (d)x(n)={1.

1-j2} (e)X(k)={16.0.1-j.1+j4. How do you linear filtering by FFT using save-add method (AU DEC 07) (AU DEC 06) 5. Compute the 8 point DFT of the sequence x(n)={0. b) Using the above signal flow graph compute DFT of x(n)=cos(n*Л)/4 . 9.0.0.0} using radix 2 DIF and DIT algorithm (AU DEC 07) 3.1+j0. Derive the equation for DIT algorithm of FFT.0.1y(n-1)+0. a) From first principles obtain the signal flow graph for computing 8 point DFT using radix 2 DIF-FFT algorithm. b) Using the above signal flow graph compute DFT of x(n)=cos(n*Л)/4 . 8.5. 0.0.5 .0.4142} 6.4142.0.1+j2. Determine the cascade and parallel form realization of the following system y(n)=-0.0.0.5.5.0<=n<=7 (AU May 07).0.6x(n-2) Expalin in detail about the round off errors in digital filters (AU DEC 04) .252x(n-2) (AU APR 05) 14.0.0.7x(n)-0.0.1+j2.2y(n-2)+3x(n)+3.1.0.0} (d)X(k)={8. Compute the IDFT of the following sequences using (a)DIT algorithm (b)DIF algorithms (a)X(k)={1.0.0.0. Draw the butterfly diagram using 8 pt DIT-FFT for the following sequences x(n)={1. Determine the direct form realization of the following system y(n)=-0.1.1-j0.4142.1-j4.0. a) From first principles obtain the signal flow graph for computing 8 point DFT using radix 2 DIT-FFT algorithm.4.1-j. State and prove circular convolution and circular conjugate properties of DFT 12.0} (c)X(k)={5.1+j.1+j.1y(n-1)+0.0. a) Discuss the properties of DFT b) Discuss the use of FFT algorithm in linear filtering 4.2.0.0.1+j.0.0} (AU May 07).0<=n<=7 10.1+j} (b)X(k)={12.1.0.72y(n-2)+0. Explain the use of FFT algorithms in linear filtering and correlation 13.0.6x(n-1)+0.4142. How do you do linear filtering by FFT using Save Add method? (AU Nov 06) 7. State and prove circular time shift and circular frequency shift properties of DFT 11.1-j2.0.

What is prewarping? 3. 20. May „07) 9. major. Find the digital filter transfer function H(Z) by using bilinear transformation method for the analog transfer function H(S)= 1/S+3 16. Distinguish between Butterworth and Chebyshev filter 2. Distinguish between the frequency response of Chebyshev Type I & Type II filter. By impulse invariant method obtain the digital filter transfer function and differential equation of the analog filter H(S)=1/S+1 (AU DEC 07) 8. Give the equation for the order of N and cut off frequency c of Butterworth filter. 22. Give the equation for converting a normalized LPF into a BPF with cutoff frequencies l and u 17. Distinguish between FIR and IIR filters (AU DEC 03) 4. . Give any two properties of Butterworth and chebyshev filters (AU DEC 07) (AU DEC 06) 5. Give the magnitude function of Butterworth filter. Write down the transfer function of the first order butterworth filter having low pass behavior (AU APR 05) 13. Distinguish between the frequency response of Chebyshev Type I filter for N odd and N even. What are the parameters(specifications) of a chebyshev filter (EC 333. Find the digital filter transfer function H(Z) by using impulse invariance method for the analog transfer function H(S)= 1/S+2 (MAY AU ‟07) 15. What are the advantages and disadvantages of bilinear transformation?(AU DEC 04) 12. 19. Determine the order of the analog butterworth filter that has a -2 dB pass band attenuation at a frequency of 20 rad/sec and atleast -10 dB stop band attenuation at 30 rad/sec (AU DEC 07) 7. Give the expression for location of poles of normalized butterworth filter (EC 333. Give the equations for the order N. Give the Butterworth filter transfer function and its magnitude characteristics for different order of filters. Give the Chebyshev filter transfer function and its magnitude response. May „07) 10. 21. What is warping effect? What is its effect on magnitude and phase response? 14. What are the properties of Chebyshev filter? (AU NOV 06). Why impulse invariance method is not preferred in the design of IIR filter other than low pass filter? 11. 25.UNIT-III IIR FILTER DESIGN PART-A 1. 26. What are the parameters that can be obtained from the Chebyshev filter specification? (AU MAY 07). (MU NOV 06). What is the effect of varying order of N on magnitude and phase response? 18. 24. minor and axis of an ellipse in case of Chebyshev filter. Give the bilinear transformation (AU DEC 03) 6. 23. Give any two properties of Butterworth low pass filters.

(AU MAY 07) Distinguish between Butterworth and Chebyshev Type I filter. 28.6x(n-1)+0. 33. Give the expression for the location of poles and zeros of a Chebyshev Type II filter. multiplications and memory locations are required to realize a system H(z) having M zeros and N poles in direct form-I and direct form –II realization? Define signal flow graph. for3/4   Assume T= 1 sec.1 sec. 41.707  H(e j 1. 46. 39.2y(n-2)+3x(n)+3. 45.1x(n-1)+0.2Л by using bilinear transformation 2. 30. cascade and parallel realization for the following Systems y(n)=-0. 37. What are the advantages and disadvantages of Bilinear transformation? What is warping effect? What is its effect on magnitude and phase response? What is Bilinear Transformation? How many numbers of additions. Apply bilinear transformation method.Determine digital Butterworth filter satisfying the following specifications: 0. Determine H(Z) for a Butterworth filter satisfying the following specifications: 0. 43. for 0  /2  H(e j 0. 42. What is the transposition theorem and transposed structure? Draw the parallel form structure of IIR filter. 35. 36. Realize the filter in mose convenient form (AU DEC 06) .6x(n-2) b) Discuss the limitation of designing an IIR filetr using impulse invariant method (AU DEC 07) 3. 38.2. Apply bilinear transformation method (AU MAY 07) 4. 34. Give the transposed direct form –II structure of IIR second order system. for 0  /4 H(e j 0. (AU APR 04) What are properties that are maintained same in the transfer of analog filter into a digital filter. What is the mapping procedure between s-plane and z-plane in the method of mapping of differentials? What is its characteristics? What is mean by Impulse invariant method of designing IIR filter? What are the different types of structures for the realization of IIR systems? Write short notes on prewarping.27.2. for /2   Assume T= 0.8  H(e j 1. Give the expression for location of poles for a Chebyshev Type I filter. 44. a) Derive bilinear transformation for an analog filter with system function H(S)=b/S+a (AU DEC 07) b) Design a single pole low pass digital IIR filter with-3 Db bandwidth of 0. What are the different types of filters based on impulse response? (AU 07) What is the most general form of IIR filter? PART B 1. Direct form II. How one can design Digital filters from Analog filters. 40. a) Obtain the direct form I. Mention any two procedures for digitizing the transfer function of an analog filter. 31. 32. 29.

Design a Chebyshev lowpass filter with the specifications p=1 dB ripple in the pass band 00. s=15 dB ripple in the stop band 0.2 7.  s =0.5 Db. Design a Butterworth low pass filter satisfying the following specifications. (AU DEC 04) p=0. Compare bilinear transformation and impulse invariant mapping 15. Briefly explain about bilinear transformation of digital filter design(AU APR 05) 13.2.1 y(n-1)+0.7x(n)-0.25x(n-2) in parallel form (EC 333 DEC „07 ) . Design a Butterworth high pass filter satisfying the following specifications. 9.4. Use impulse invariance to obtain H(Z) if T= 1 sec and H(s) is 1/(s3 +3s2 +4s+1) 1/(s2+2 s +1) 11.p=0.5. b)Realize the system given by difference equation y(n)=-0.15 HZ. a) Design a chebyshev filter with a maxmimum pass band attenuation of 2.2 (AU APR 04) 14. Use bilinear transformation method to obtain H(Z) if T= 1 sec and H(s) is 1/(s+1)(S+2) (AU DEC 03) 2 1/(s +2 s +1) 12. Design a digital filter equivalent to this using impulse invariant method H(S)=10/S2+7S+10 (AU DEC 03)(AU DEC 04) 10. at Ωp=20 rad/sec and the stop band attenuation of 30 Db at Ωs=50 rad/sec.5 dB s=0. Use bilinear transform to design a butterworth LPF with 3 dB cutoff frequeny of 0. p =1 dB.s=15 dB:F=1Hz. s=15 dB  p =0. Design (a) a Butterworth and (b) a Chebyshev analog high pass filter that will pass all radian frequencies greater than 200 rad/sec with no more that 2 dB attuenuation and have a stopband attenuation of greater than 20 dB for all  less than 100 rad/sec.3  using impulse invariance method(AU DEC 06) 6.72y(n-2)+0. 8.10 Hz.

UNIT IV FIR FILTER DESIGN PART A 1. 22. (MU 02) What are desirable characteristics of windows? What is the principle of designing FIR filters using windows? What is a window and why it is necessary? Draw the frequency response of N point rectangular window. 7. 16. 23. 2. 20. Also discuss symmetric and antisymmetric cases of FIR filter (AU DEC 07) 3. (MU Nov 03) Give the equation specifying Kaiser window.1. 3. (AU „07) 2. n=0. 9. 8. (MU 03) Give the equation specifying Hanning and Blackman windows. 10. Give the expression for the frequency response of Draw the frequency response of N point Bartlett window Draw the frequency response of N point Blackman window Draw the frequency response of N point Hanning window. 0. 6. Compare rectangular and hanning window functions Briefly explain the frequency sampling method of filter design Compare frequency sampling and windowing method of filter design PART-B 1. Use window method with a Hamming window to design a 13-tap differentiator (N=13). DEC 03) . 18. (AU DEC 03) What is the necessary and sufficient condition for linear phase characteristics in FIR filter. 24. 11. (AU 06) What is Gibbs phenomenon? (AU DEC 04) (AU DEC 07) Show that the filter with h(n)={-1. 13. What are the issues in designing FIR filter using window method?(AU APR 04. 4. 15. What are the desirable and undesirable features of FIR filter? Discuss the stability of the FIR filters (AU APR 04) (AU DEC 03) What are the main advantages of FIR over IIR (AU APR 04) What is the condition satisfied by Linear phase FIR filter? (DEC 04) (EC 333 MAY 07) What are the design techniques of designing FIR filters? What condition on the FIR sequence h(n) are to be imposed in order that this filter can be called a Linear phase filter? (AU 07) State the condition for a digital filter to be a causal and stable. i) Prove that FIR filter has linear phase if the unit impulse responsesatisfies the condition h(n)=h(N-1-n). 12. 14. 1} is a linear phase filter Explain the procedure for designing FIR filters using windows. 17. 5. 19. 21.……M-1.

Design an approximation to an ideal bandpass filter with magnitude response H(ej) = 1 .75 0 for 0. Plot the frequency response in both Cases 11. Describe the window sequences generally used and compare their properties 5. Derive the frequency response of a linear phase FIR filter when impulse responses symmetric & order N is EVEN and mention its applications 6. 434 0 . (AU DEC 04) 8.75  by using rectangular window function of length N=11.25   0.1. b) Determine the coefficients of a linear phase FIR filter of length N=15 which has Symmetric unit sample response and a frequency response that satisfies the following conditions H r (2 k/15) = 1 for k=0.25 and 0. (AU DEC 07) 10. i) Explain the type I design of FIR filter using frequency sampling method ii) A low pass filter has the desired response as given below Hd(ej)= e –j3.3 .2.5 k=5 = 0. Design an ideal band pass digital FIR filter with desired frequency response H(e j )= 1 for 0. i) Derive the frequency response of a linear phase FIR filter when impulse responses antisymmetric & order N is odd ii) Explain design of FIR filter by frequency sampling technique (AU MAY 07) 7.4. Design an Ideal Hilbert transformer using hanning window and Blackman window for N=11. Design a 15-tap linear phase filter to the following discrete frequency response (N=15) using frequency sampling method (MU 03) H(k) = 1 0k4 = 0. a) How is the design of linear phase FIR filter done by frequency sampling method? Explain. otherwise Take N=11.25 k=6 = 0. ii) Explain the need for the use of window sequences in the design of FIR filter. 0≤≤Л/2 0 Л/2≤≤Л Determine the filter coefficients h(n) for M=7 using frequency sampling technique (AU DEC 07) 7.1 k=7 =0 elsewhere 9.

An FIR filter is given by the difference equation y(n)=2x(n)+4/5 x(n-1)+3/2 x(n-2)+2/3 x(n-3) Determine its lattice form(EC 333 DEC 07) 13. 2‟s complement and 1‟s complement (AU DEC 06) 3. Brief on co-efficient inaccuracy. What is meant by A/D conversion mode? 18. Express the fraction 7/8 and -7/8 in sign magnitude.Use hanning and hamming window (AU DEC 04) UNIT V FINITE WORD LENGTH EFFECTS PART –A 1. What is product quantization error? 17. The length of the impulse response should be 7. What is the effect of quantization on pole locations? .0 for k=4 0 for k=5. What are the different quantization methods? (AU DEC 06) (AU DEC 07) 5. What do you understand by sign-magnitude representation? 7. What are the three quantization errors due to finite word length registers in digital filters? (MU Oct‟98) 13. What do you understand by input quantization error? 16. Design an Ideal Hilbert transformer using rectangular window and Black man window for N=11.( EC 333 DEC 07) 16. 04 0 . Using a rectangular window technique design a low pass filter with pass band gain of unity cut off frequency of 1000 Hz and working at a sampling frequency of 5 KHz. (MU Oct‟96) 12. otherwise Take N=11. What do you understand by a fixed point number? (MU Oct‟95) 2. What do you understand by 2‟s complement representation? 8. What is meant by block floating point representation? What are its advantages? 10. What are the quantization errors due to finite word length registers in digital filters? 4. what are advantages of floating point arithmetic? 11. Compare the fixed point and floating point arithmetic. What are the different types of fixed point number representation? 6. 15. How the multiplication and addition are carried out in floating point arithmetic? 14.6. Plot the frequency response in both Cases (EC 333 DEC ‟07) 9. Write an account on floating point arithmetic? (MU Apr 2000) 9.7 12. Design an approximation to an ideal lowpass filter with magnitude response H(ej) = 1 . 17.

Find the quantization step size of the quantizer with 3 bits 31. May „07) 34.6 and find out the output round off noise power. Assume a word length of 4-bits through truncation. Give the expression for signal to quantization noise ratio and calculate the improvement with an increase of 2 bits to the existing bit. Assume a1=0. Express the following binary numbers in decimal A) (100111. Explain the characteristics of Limit cycle oscillations with respect to the system described by the differential equations. (ECE AU‟ 04)(EC 333 DEC 07) 3.Find the output round off noise power.U.U. Apr 00) 31.9z-1+0. Explain briefly the need for scaling digital filter implementation? (M. What is zero input limit cycle overflow oscillation (AU 07) 21. Draw the quantization noise model for a first order IIR system 27. What is meant by rounding? Draw the pdf of round off error 28. Determine the dead band of the filter with pole at 0. Apr 2000) (AU DEC 07) 29. i) Describe the quantization errors that occur in rounding and truncation in two‟s complement ii) Draw a sample/hold circuit and explain its operation iii) What is a vocoder? Expalin with a block diagram (AU DEC 07) 6. What is meant by truncation? Draw the pdf of round off error 29. H2(z)=1/(1-a2z-2). What are the assumptions made concerning the statistical independence of various noise sources that occur in realizing the filter? (M.19.U Oct 97. Consider the transfer function H(z)=H1(z)H2(z) where H1(z)=1/(1-a1z-1) . y(n)=0. 32.5 and a2=0.1110)2 (B) (101110. Define the deadband of the filter? (AU 06) 25. Find the effect of coefficient quantiztion on pole locations of the given second order IIR system when it is realized in direct form –I and in cascade form. 98.011)2 33.8Z-1) (AU DEC 07) . Draw the quantization noise model for a second order system and explain H(z)=1/(1-2rcosz-1+r2z-2) and find its steady state output noise variance (ECE AU‟ 05) 2. List the different types of frequency domain coding 35.U Oct 98)(AU-DEC 07) 30.5 and the number of bits used for quantization is 4(including sign bit) 26. Two first order low pass filter whose system functions are given below are connected in cascade.Why rounding is preferred to truncation in realizing digital filter? (EC 333. What is subband coding? PART-B (EC 333 MAY 07) (EC 333 MAY 07) 1. What do you mean by quantization step size? 30. Determine the overall output noise power H1(Z)=1/(1-0.95y(n-1)+x(n) and determine the dead band of the filter (AU‟ Nov 04) 5. What is meant by limit cycle oscillations?(M. H(z)= 1/(1-0. Why rounding is preferred than truncation in realizing digital filter? (M.9Z-1) H2(Z)=1/(1-0.1111)2 C (10011. Apr 96) 20.2z –2) (AU‟ Nov 05) 4.

1875 in One‟s complement form sign magnitude form Two‟s complement form. Compute the pole positions in z-plane and calculate the scale factor So to prevent overflow in adder 1.52y (n-1) + x (n) 13.999y(n-1)+x(n) is applied to an ADC. Study the limit cycle behavior of the following systems i.05( 1+z-1)2 /(1-1.65y(n-2) + 0.7y(n-1) + x (n) ii. 8. Derive the truncation error and round off error noise power and compare both errors 17. 19. Convert the following decimal numbers into binary: 1) (20. The input to the system y(n)=0. Consider a Butterworth lowpass filter whose transfer function is H(z)=0. For the system with system function H (z) =1+0.78125 and -0. Derive the quantization input nose power and determine the signal to noise ratio of the system 16.1875 C) 225.4z-1 draw the signal flow graph 14.3275 10.8 z-2 ). Express the following decimal numbers in binary form A) 525 B) 152. Find the steady state variance of the noise in the output due to quantization of input for the (EC 333 DEC 07) first order filter y(n)=ay(n-1)+x(n) .75z-1 / 1-0. 11. y(n) = 0.675) 10 2) (120. Express the decimal values -6/8 and 9/8 in (i) Sign magnitude form (ii) One‟s complement form (iii) Two‟s complement form 12. Derive the scaling factor So that prevents the overflow limit cycle oscillations in a second order IIR system.7. Express the decimal values 0. What is the power produced by the quantization noise at the output of the filter if the input is quantized to 1) 8 bits 2) 16 bits (EC 333 DEC 07) (EC 333 DEC 07) 19. Explain product quantization error and coefficient quantization error with examples 18.75) 10 20. y(n) = 0. and find scale factor s0 to prevent overflow limit cycle oscillations 15.2z-1 +0.

SIGNALS AND SYSTEMS PART A 1.01 πn Wo=0.01πn (b) sin (π62n/10) a) Cos 0. Determine which of the following sinusoids are periodic and compute their fundamental period (a) Cos 0. and find Nyquist rate of the signal x(t)=5 sin250 t + 6cos300 t April/May2008 CSE A band limited continuous time signal.ANAND INSTITUTE OF HIGHER TECHNOLOGY KAZHIPATTUR.Therefore the signal is periodic Fundamental period N=2π [m/wo] =2π(m/0.01 π the fundamental frequency is multiply of π . State sampling theorem .Therefore the signal is periodic Fundamental period N=2π [m/wo] =2π(m/(π62/10)) Choose the smallest value of m that will make N an integer M=31 N=2π(310/62π) N=10 Fundamental period N=10 2. State sampling theorem Nov/Dec 2008 CSE A band limited continuous time signal.1 N=2π(0.01π) N=20 Fundamental period N=20 b) sin (π62n/10) Wo=0.01 π the fundamental frequency is multiply of π . with higher frequency f max Hz can be uniquely recovered from it‟s samples provided that the sampling rate Fs>2f max samples per second.01π) Choose the smallest value of m that will make N an integer M=0.1/0.J Class : VII Sem/CSE A&B UNIT-1 . CHENNAI –603 103 DEPARTMENT OF ECE Date: 15-05-2009 PART-A QUESTIONS AND ANSWERS Subject : Digital signal Processing Sub Code : IT1252 Staff Name: Robert Theivadas. with higher frequency f max Hz can be uniquely recovered from its samples provided that the sampling rate Fs>2fmax samples per second 3. Nyquist rate Nov/Dec 2008 CSE .

Therefore. State and prove convolution property of Z transform. the signal is Periodic . Determine which of the following signals are periodic and compute their fundamental period. April/May2008 Convolution Property (MAY 2006 ECESS) CSE 5.The Fundamental frequency is multiply of п.x(t)=5 sin250t+ 6cos300 t Frequency present in the signals F1=125Hz F2=150Hz Fmax=150Hz Fs>2Fmax=300 Hz The Nyquist rate is FN= 300Hz 4. 5п .Therefore. Fundamental period N=2п [m/wo] = 2п [m/√2п] m=√2 =2п [√2/√2п] N=2 (b) sin 20пt + sin 5пt wo=20п.The Fundamental frequency is multiply of п. Fundamental period of signal sin 20пt N1=2п [m/wo] =2п [m/20п] m=1 =1/10 Fundamental period of signal sin 5пt N2=2п [m/wo] =2п [m/5п] m=1 =2/5 N1/N2=(1/10)/(2/5) =1/4 4N1=N2 . the signal is Periodic . Nov/Dec 2007 CSE (a) sin √2пt (b) sin 20пt + sin 5пt (a) sin √2пt wo=√2п .

Nov/Dec 2007 CSE Soln: x1(n)={1.2.1} and x2(n)={4.2.3. Define Z transform for x(n)=-nan u(-n-1) X(n) =-nan u(-1-n) X (z)= = = == -z d/dz X(z) =z d/dz( )= u(-n-1)=0for n>1 8.3.1}.16.2.3.1} x2(n)={4.1}.15 April/May 2008 IT 7. Determine the circular convolution of the sequence x1(n)={1.N= 4N1=N2 N=2/5 6.3. Y(n)= 15.21.2. Find whether the signal y= n2 x(n) is linear Y= x(n) Y1(n)=T[x1(n)]= Y2(n)= T[x2(n)]= x1(n) x2(n) April/May 2008 IT .

0.2.3} 13. 12. output and input respectively. Output. 14.0. for n≥0 = 0 for n<0 11. Is the system y(n)=ln[x(n)] is linear and time invariant? (MAY 2006 IT) The system y(n)=ln[x(n)] is non-linear and time invariant alnx1(n)+blnx2(n) ≠ ln(ax1(n)+bx2(n)  Non-linear system lnx (n)=lnx (n-n0)  Time invariant system 10. What are the different methods of evaluating inverse z transform. n=0 =0.1.  DT sinusoid whose frequencies are separated by an integer multiple of 2π are identical. (NOV 2005 IT)  DT sinusoid is periodic only if its frequency f is a rational number.3} y(n) = x(n-1) ={3. 15.1. Discrete time unit impulse function δ(n) =1. Input and response of the system respectively. (APR 2004 IT) H(z)=Y(z)/X(z) H(z).2. Define system function and stability of a DT system. List the properties of DT sinusoids.1. Write down the expression for discrete time unit impulse and unit step function. * represent the convolution operator y(n). (APR 2005 IT).1. x(n)&h(n). Determine the response a system with y(n)=x(n-1) for the input signal x(n) = |n| for -3≤n≤3 = 0 otherwise (NOV 2005 IT) x(n)= {3. Find out the DFT of the signal X(n)= (n) Nov/Dec 2008 CSE .Y(z) & X(z)z-transform of the system impulse.2. n≠0 Discrete time step impulse function. (NOV 2004 IT)  Long division method  Partial fraction expansion method  Residue method  Convolution method UNIT-II . What is the causality condition for an LTI system? (NOV 2004 IT) Conditions for the causality h(n)=0 for n<0 16.FAST FOURIER TRANSFORMS 1. u(n) = 1. Define linear convolution of two DT signals.2. (APR 2004 IT) y(n)=x(n)*h(n).The weighted sum of input is a1 T[x1(n)]+a2 T[x2(n)]=a1 x1(n)+a2 x2(n)-----------1 the output due to weighted sum of input is y3(n)=T[a1X1(n)+a2X2(n)] = a1 x1(n)+a2 x2(n)----------------------------------2 9.

1. so bit-reversed indexes are used to combine FFT stages. Therefore the MSB's become LSB's and the LSB's become MSB's.1} 2.The data ordering required by radix-2 FFT's turns out to be in "bit reversed" order. Draw radix 4 butterfly structure for (DIT) FFT algorithm April/May2008 CSE .0} X(k)={1. What is meant by bit reversal and in place commutation as applied to FFT? Nov/Dec 2008 CSE "Bit reversal" is just what it sounds like: reversing the bits in a binary word from left to write. Input sample index 0 1 2 3 4 5 6 7 Binary Representation 000 001 010 011 100 101 110 111 Bit reversed binary 000 100 010 110 001 101 011 111 Bit reversal sample index 0 4 2 6 1 5 3 7 3.X(n)={1.0.0.1.

Find DFT for {1.1}. Draw the basic butterfly diagram for radix 2 DIT-FFT and DIF-FFT. Nov/Dec 2007 CSE Butterfly Structure for DIT FFT The DIT structure can be expressed as a butterfly diagram MAY 2006 ECESS &(NOV 2006 ITSS) .4.0. 2008 IT April/May2008 CSE /April/May 5.0.

Differentiate between DIT and DIF FFT algorithms. What are the advantages of Bilinear mapping April/May 2008 IT  Aliasing is avoided  Mapping the S plane to the Z plane is one to one  The closed left half of the S plane is mapped onto the unit disk of the Z plane 7. Processing speed is very high compared to the direct computation of DFT. Define Complex Conjugate of DFT property. How may multiplication and addition is needed for radix-2 FFT? April/May 2008 IT Number of complex addition is given by N Number of complex multiplication is given by N/2 8.What are the advantages of FFT algorithm over direct computation of DFT? (May/June 2007)-ECE The complex multiplication in the FFT algorithm is reduced by (N/2) log2N times.The DIF structure expressed as a butterfly diagram 6. Define DTFT pair? (May/June 2007)-ECE The DTFT pairs are (MAY 2006 IT) -j2πkn/N X(k) = x(n)e X(n) = x(k)ej2πkn/N 9.No 1 DIT FFT algorithm Decimation in time FFT algorithm . (MAY 2006 IT) DIF FFT algorithm Decimation in frequency FFT algorithm 2 Twiddle factor k=(Nt/2m) Twiddle factor k=(Nt/2M-m+1) 11. (May/Jun 2007)-ECE DFT If x(n)↔X(k) then N X*(n)↔(X*(-k))N = X*(N.K) 10. S.Give any two properties of DFT (APR 2004 IT SS) Linearity : DFT [ax(n)+b y(n)]=a X(K)+bX(K) Periodicity: x(n+N)=x(n) for all n X(K+N)=X(K) for all n 12.

It is based on fundamental principal of decomposing the computation of DFT of a sequence of the length N in to successively smaller discrete Fourier Transforms. (APR 2005 IT) FFT algorithms. Then state the reason (APR 2005 ITDSP) DFT is a finite length sequence periodic. What are the requirements for converting a stable analog filter into a stable digital filter? Nov/Dec 2008 CSE  The JΩ axis in the s plane should be map into the unit circle in the Z plane . What do you mean by in place computation in FFT. Distinguish between the frequency response of chebyshev type I and Type II filter Nov/Dec 2008 CSE . What is FFT? (Nov/Dec 2006)ECE The fast Fourier transform is an algorithm is used to calculate the DFT.Is the DFT is a finite length sequence periodic.Determine the DIFT of a sequence x(n) = an u(n) (Nov/Dec 2006)-ECE X(K) = x(n) ej2πkn/N The given sequence x(n) = an u(n) DTFT{x(n)} = x(n) ej2πkn/N j2πk/N n = (a e ) Where an = 1-an/(1-a) X(K) = (1 – aNej2πk)/ (1-aej2πk/N) 15. UNIT-III .13. with period 2π.thus the stable analog filter will be converted to a stable digital filter 2. The FFT algorithm provides speed increase factor when compared with direct computation of the DFT.IIR FILTER DESIGN 1. 14. for computing the DFT when the size N is a power of 2 and when it is a power of 4 16.thus there will be a direct relationship between the two frequency variables in the two domains  The left half plane of the s plane should be map into the inside of the unit circle in the z – plane . N-1 X(ej )= Σ x(n) e-jn n =0 j X(e ) is continuous & periodic in .

ΏC (s2 + Ώ1 Ώ2) / s (Ώ2 .Write frequency translation for BPF from LPF April/May2008 CSE Low pass with cut – off frequency ΏC to band –pass with lower cut-off frequency Ώ1 and higher cut-off frequency Ώ2: S ------------. What is the need for prewraping in the design of IIR filter Nov/Dec 2008 CSE The warping effect can be eliminated by prewarping the analog filter .Ώ1)} 5. as frequency Chebyshev filters.Type I chebyshev filter Type II chebyshev filter Type I chebyshev filters are all pole filters that exhibit equirpple behavior in the pass band and monotonic in stop band .Ώ1) The system function of the high pass filter is then H(s) = Hp { ΏC ( s2 + Ώ1 Ώ2) / s (Ώ2 . Chebyshev filters CSE Butter Worth Filter Magnitude response of Butterworth filter decreases monotonically.Type II chebyshev filters contain both poles and zeros and exhibits a monotonic behavior in the pass band and an equiripple behavior in the stop band 3. Magnitude response of chebyshev filter exhibits ripple in pass band April/May2008 .This can be done by finding prewarping analog frequencies using the formula Ω = 2tan-1ΩT/2 4.Compare Butterworth.

Linear Linear phase response for phase is obtained if h(n) = ± h(m-1-n) H(z) = ±Z-1H(Z-1) These filters are to be designed for These are inherently stable stability filters . obtain the digital filter transfer function and differential equation of the analog filter H(s)=1 / (s+1) Nov/Dec 2007 CSE H(s) =1/(s+1) Using partial fraction H(s) =A/(s+1) = 1/(s-(-1) Using impulse invariance method H (z) =1/1-e-Tz-1 AssumeT=1sec H(z)=1/1-e-1z-1 H(z)=1/1-0. They are feedback.1 αs -1/ 100. These filters use feedback These filters do not use from output. Ωs = 30 rad/sec log√100.1 αp -1 N≥ Log αs/ αp log√10 -1/ 100. By Impulse Invariant method. Ωp =20 rad/sec αs = 10 dB. Determine the order of the analog Butterworth filter that has a -2 db pass band attenuation at a frequency of 20 rad/sec and atleast -10 db stop band attenuation at 30 rad/sec. recursive filters. Sl.Distinguish between FIR and IIR filters. present. Nov/Dec 2007CSE αp =2 dB. Nonlinear phase response.2 -1 N≥ Log 30/ 20 ≥3.No 1 2 Nov/Dec 2007 CSE 3 4 5 IIR FIR H(n) is infinite duration H(n) is finite duration Poles as well as zeros are These are all zero filters. They are nonrecursive. Sometimes all pole filters are also designed.3678z-1 8.37 Rounding we get N=4 7.increases from 0 ∞ Poles of the chebyshev filter lies on the ellipse Poles on the butter worth lies on the circle 6.

Many to one mapping. This is undone by pre-warping the critical frequencies of the analog filter (cutoff frequency. e.then = 1/2∏j 10. The magnitude response of the Butterworth filter closely approximates the ideal response as the order N increases.What is frequency warping? (MAY 2006 IT DSP) The bilinear transform is a method of compressing the infinite. Give any two properties of Butterworth filter and chebyshev filter.Define Parsevals relation April/May 2008 IT If X1(n) and X2(n) are complex valued sequences . The magnitude response of the chebyshev type-I filter exhibits ripple in the pass band. 12. (APR 2005 ITDSP) The FIR system is a non-recursive system. Impulse invariance. The poles of the Chebyshev type-I filter lies on an ellipse.. S = (2/T) (Z-1) (Z+1) 13. Windowing 2. straight analog frequency axis to a finite one long enough to wrap around the unit circle only once. linear frequency relationship between analog and its transformed digital frequency. Bilinear Transform 1. 7 More complexity of implementation Less complexity of implementation 8 Less memory is required More memory is requied 9 Design procedure is complication Less complicated 10 Design methods: Design methods: 1. described by the difference equation M-1 6 . with minimum order are required 9. 2. The magnitude response of the Butterworth filter decreases monotonically as the frequency increases (Ώ) from 0 to ∞.What are the advantages and disadvantages of bilinear transformation? (May/June 2006)-ECE Advantages: 1. the designed digital filter will meet the desired specifications. d. This is also sometimes called frequency warping. This introduces a distortion in the frequency. The poles on the Butterworth filter lies on the circle. center frequency) such that when the analog filter is transformed into the digital filter. b.Find the transfer function for normalized Butterworth filter of order 1 by determining the pole values. 2. Frequency sampling 11 Can be used where sharp Used where linear phase cutoff characteristics characteristic is essential.Differentiate between recursive and non-recursive difference equations.Number of More multiplication requirement is less. c. (MAY 2006 IT DSP) Poles = 2N N=1 Poles = 2 14. Disadvantage: Aliasing 11. (Nov/Dec 2006) a.

9976 N = (log(99.995 ε = (100.995/0.Write procedure for designing FIR filter using windows. April/May2008 CSE The oscillatory behavior of the approximation XN(W) to the function X(w) at a point of discontinuity of X(w) is called Gibb‟s Phenomenon 2.What is gibb’s Phenomenon. (ii) Cascade structure realization.Mention advantages of direct form II and cascade structures.Give the bilinear transformation. multiplied by T. 19. Each sum system in the cascade structure is realized in direct form-II.1αs-1)1/2 = 99. Ω = 2tan-1ΩT/2 This equation shows that frequency relationship is highly nonlinear.Find the order and poles of Butterworth LPF that has -3dB bandwidth of 500 Hz and an attenuation of -40 dB at 1000 Hz. (NOV 2005 ITDSP) αp = -3dB αs = -40dB Ωs = 1000*2π rad/sec Ωp=500*2π The order of the filter N ≥(log(λ/ε))/(log(Ωs/Ωp)) λ = (100. (APR 2004 ITDSP) (i) The main advantage direct form-II structure realization is that the number of delay elements is reduced by half. The order of each sub system may be two or three (depends) or more. 18. Begin with the desired frequency response specification Hd(w) . The specifications of equivalent analog filter are obtained by following relationship.64 N ≥ 6. the system function is expressed as a product of several sub system functions.9976))/(log(2000π/1000π)) = 2/0.This procedure involves choosing the response of the digital filter as an equi-spaced sampled version of the analog filter. 17. straight analog frequency axis to a finite one long enough to wrap around the unit circle only once.Σ aky(n-k) k=0 k=1 15. What is prewarping? (Nov/Dec 2003)-ECE When bilinear transformation is applied. UNIT-IV . (Nov/Dec 2003)-ECE The bilinear transformation method overcomes the effect of aliasing that is caused due to the analog frequency response containing components at or beyond the nyquist frequency.What is impulse invariant mapping? What is its limitation? (Apr/May 2005)-ECE The philosophy of this technique is to transform an analog prototype filter into an IIR discrete time filter whose impulse response [h(n)] is a sampled version of the analog filter‟s impulse response.FIR FILTER DESIGN 1.3 = 6.y(n) = Σ bkx(n-k) k=0 The IIR system is a non-recursive system. It is also called frequency warping. Ω = 2/T tan ω/2 This is called prewarping relationship.1αp-1)1/2 = 0. the discrete time frequency is related continuous time frequency as. Hence. April/May2008 CSE 1. The bilinear transform is a method of compressing the infinite.64 = 7 Poles=2N=14 16. This effect can be nullified by applying prewarping. described by the difference equation N M y(n) = Σ bkx(n-k). the system complexity drastically reduces the number of memory elements .

the output signal lagging in phase but the frequency remain the same Y(n)= A .Let the input signal be X(n)=A .What are Gibbs oscillations? Nov/Dec 2007 CSE Oscillatory behavior observed when a square wave is reconstructed from finite number of harmonics. the signal level at certain points in the digital filters must be scaled so that no overflow occur in the adder 5. In This equation that the output is the time delayed signal and is more commonly known as phase delayed at w=wo Is called phase delay 7. The unit cell of the square wave is given by Its Fourier series representation is 4.FIR filter can be realized in both recursive and non recursive structure 4.Filters wit h any arbitrary magnitude response can be tackled using FIR sequency 6.FIR filter has exact linear phase 2. Indeed hd(n) is related to Hd(w) by the Fourier Transform relation. 3. (MAY 2006 IT DSP) & (NOV 2004 ECEDSP) . Explain briefly the need for scaling in the digital filter realization Nov/Dec 2007 CSE To prevent overflow. Determine the corresponding unit sample response hd(n) 3. Define Phase Dealy April/May 2008 IT When the input signal X(n) is applied which has non zero response the output signal y(n) experience a delay with respect to the input signal . What are the advantages of FIR filters? April/May 2008 IT 1. State the advantages and disadvantages of FIR filter over IIR filter. + Where A= Maximum Amplitude of the signal Wo=Frequency in radians f=phase angle Due to the delay in the system response .FIR filter always stable 3.2.

symmetric  Linear phase frequency response  Impulse invariance 14. What are the desirable and undesirable features of FIR Filters? (May/June 2006)ECE The width of the main lobe should be small and it should contain as much of total energy as possible. List the important characteristics of physically realizable filters. 0≤n≤M-1 0.42 + 0. 0≤n≤M-1 = 0. What is the condition for linear phase of a digital filter? (APR 2005 ITDSP) h(n) = h(M-1-n) Linear phase FIR filter with a nonzero response at ω=0 h(n) = -h(M-1-n)Low pass Linear phase FIR filter with a nonzero response at ω=0 12.5 cos2πn/ (M-1) +0.5 cos2πn/ (M-1) +0.08 cos4πn/(M-1). 8.Direct form-I 2. Define backward and forward predictions in FIR lattice filter. otherwise The window function of a non causal Blackman window is expressed by WB(n) = 0.5cos2πn/ (M-1). otherwise The width of the main lobe is approximately 12π/M and the peak of the first side lobe is at -58dB.The side lobes should decease in energy rapidly as w tends to π 10. . Otherwise The width of the main lobe is approximately 8π/M and thee peak of the first side lobe is at -32dB. 11.5 – 0.5 + 0.….  It can be realized with recursive and non-recursive structures  It is free of limit cycle oscillations when implemented on a finite word length digital system Disadvantages of FIR filter over IIR filter  Obtaining narrow transition band is more complex. (NOV 2005 ITDSP) Symmetric and anti.Advantages of FIR filter over IIR filter  It is a stable filter  It exhibit linear phase. hence can be easily designed. 0≤|n|≤(M-1)/2 = 0.2. Direct form-II 9. 0≤|n|≤(M-1)/2 0. Define Hanning and Blackman window functions. Write the magnitude function of Butterworth filter.3.  Memory requirement is very high  Execution time in processor implementation is very high. What is the effect of varying order of N on magnitude and phase response? (Nov/Dec2005) -ECE |H(jΏ)|2 = 1 / [ 1 + (Ώ/ΏC)2N] where N= 1. The window function of a causal Blackman window is expressed by WB(n) = 0.42 – 0. (NOV 2005 IT) The reflection coefficient in the lattice predictor is the negative of the cross correlation coefficients between forward and backward prediction errors in the lattice. Otherwise The window function of non-causal Hanning window I s expressed by WHann(n) = 0. (May/June 2006)-ECE The window function of a causal hanning window is given by WHann(n) = 0. List out the different forms of structural realization available for realizing a FIR system.08 cos4πn/(M-1).5cos2πn/ (M-1). 13. (MAY 2006 IT DSP) The different types of structures for realization of FIR system are 1.

(NOV 2004 ITDSP) The transfer function of Barlett window wB(n) = 1-(2|n|)/(N-1). it is in stable. ((N-1)/2)≥n≥-((N-1)/2) α = 0. for |n| ≤ (M-1)/2 Where α is an independent variable determined by Kaiser. The desired frequency response of a FIR filter can be represented as ∞ Hd(ejω)= Σ hd(n)e-jωn n= -∞ If h(n) is absolutely summable(i. Some of the most frequently used window functions are described in the following sections 16. What is meant by FIR filter? And why is it stable? (APR 2004 ITDSP) FIR filter  Finite Impulse Response.. Mention two transformations to digitize an analog filter.e.Give the equation specifying Barlett and hamming window. Β = α[ 1 – (2n/M-1)2] 17. List the characteristics of FIR filters designed using window functions. Give the Kaiser Window function. Nov/Dec 2008 CSE&MAY 2006 IT . (APR 2004 ITDSP) (i) Impulse-Invariant transformation techniques (ii) Bilinear transformation techniques 19. (Apr/May 2004)-ECE The Kaiser Window function is given by WK(n) = I0(β) / I0(α) .54+0. So. (NOV 2004 ITDSP)  20. Compare fixed point and floating point arithmetic.54 UNIT-V . Bounded Input Bounded Output Stable).FINITE WORD LENGTH EFFECTS 1. ((N-1)/2)≥n≥-((N-1)/2) The transfer function of Hamming window whm(n) = 0. Draw the direct form realization of FIR system.46cos((2πn)/(N-1).15. 18. NOV 2004 ITDSP the Fourier transform of the window function W(ejw) should have a small width of main lobe containing as much of the total energy as possible  the fourier transform of the window function W(ejw) should have side lobes that decrease in energy rapidly as w to π.

What are the effects of truncating an infinite flourier series into a finite series? Nov/Dec 2008 CSE 4.List errors due to finite world length in filter design CSE    Input quantization error Product quantization error Coefficient quantization error April/May2008 .Truncation error Et=Nt-N 3. Compared to FPA. accuracy is poor Compared to FPA it is low cost and easy to design It is preferred for real time operation system Errors occurs only for multiplication Processing speed is high Overflow is rare phenomenon Floating Point Arithmetic  It covers a large range of numbers  It attains its higher accuracy  Hardware implementation is costlier and difficult to design  It is not preferred for real time operations.  Truncation and rounding errors occur both for multiplication and addition  Processing speed is low  Overflow is a range phenomenon 2.Quantization error 2.What are the errors that arise due to truncation in floating point numbers Nov/Dec 2008 CSE 1.Fixed Point Arithmetic        It covers only the dynamic range. Draw block diagram to convert a 500 m/s signal to 2500 m/s signal and state the problem due to this conversion April/May2008 CSE 5.

What do you mean by limit cycle oscillations in digital filter? Nov/Dec 2007 CSE In recursive system the nonlinearities due to the finite precision arithmetic operations often cause periodic oscillations to occur in the output .even when the input sequence is zero or some non zero constant value .02b-20log10 (RFS /ζx) dB.What are the different quantization methods? (Nov/Dec 2006)-ECE  amplitude quantization  vector quantization  scalar quantization 12. Thus oscillation is called as an overflow limit cycle oscillation.81+6. 10.e. up-sampling.Give the expression for signal to quantization noise ratio and calculate the improvement with an increase of 2 bits to the existing bit.List the advantages of floating point arithmetic. which are stored at the registers.. 15.. What is truncation error? (APR 2005 ITDSP) Truncation is an approximation scheme wherein the rounded number or digits after the pre-defined decimal position are discarded. i. What is meant by overflow limit cycle oscillations? (May/Jun 2006 ) In fixed point addition. down-sampling.Zero input limit cycle oscillation ii.5. What are decimators and interpolators? (APR 2005 ITDSP) Decimation is a process of reducing the sampling rate by a factor D.e.2-b for x >0 and 0≤ < -2. With b = 2 bits increase. What are the types of limit cycle oscillation? April/May 2008 IT i. 11.if the mantissa is truncated to b bits . 14.What is the effect of down sampling on the spectrum of a signal? (APR 2005 ITDSP) & (APR 2005 ITDSP) .2-b for x <0 8. the signal to noise ratio will increase by 6. overflow occurs due to excess of results bit.Define truncation error for sign magnitude representation and for 2’s complement Representation April/May 2008 IT&APR 2005 IT Truncation is a process of discarding all bits less significant than least significant bit that is retained For truncation in floating point system the effect is seen only in mantissa. Nov/Dec 2005)-ECE SNRA / D = 16. 16. Interpolation is a process of increasing the sampling rate by a factor I. oscillation will occur in the system. (Nov/Dec 2006. Due to this overflow.such oscillation in recursive system are called limit cycle oscillation 7. i.then the error satisfies 0≥ > -2. How will you avoid Limit cycle oscillations due to overflow in addition(MAY 2006 IT DSP) Condition to avoid the Limit cycle oscillations due to overflow in addition |a1|+|a2|<1 a1 and a2 are the parameter for stable filter from stability triangle.02 X 2 = 12dB. (Nov/Dec 2006)-ECE  Large dynamic range  Occurrence of overflow is very rare  Higher accuracy 13.overflow limit cycle oscillation 9.

17. In the case of speech signal processing. (APR 2004 ITDSP) & (NOV 2003 ECEDSP) & (NOV 2005 ECEDSP) Sub band coding of speech is a method by which the speech signal is subdivided into several frequency bands and each band is digitally encode separately. |ω|≤ л/M = 0. downsampling. since it involves the magnitude of the number. 19. The quantized error εR due to rounding is given by εR=QR(x)-x where QR(x) = quantized number(rounding error) The rounding error is independent of the types of fixed point arithmetic. Thus. The rounding error is symmetric about zero and falls in the range. i. -((2-bT-2-b)/2)≤ εR ≤((2-bT-2-b)/2) εR may be +ve or –ve and depends on the value of x. (APR 2004 ITDSP) The basic operations in multirate signal processing are (i)Decimation (ii)Interpolation Decimation is a process of reducing the sampling rate by a factor D.The signal (n) with spectrum X(ω) is to be down sampled by the factor D. An antialiasing filter eliminates the spectrum of X(ω) in the range (л/D≤ ω ≤π. What is the output? (NOV 2004 ITDSP) .What is an anti-imaging filter? (NOV 2004 ITDSP) The image signal is due to the aliasing effect..Define the basic operations in multirate signal processing. there will be M-1 additional images of the input spectrum.e.0.2-bR/2 18.-1. Define sub band coding of speech.What is the effect of quantization on pole locations? (NOV 2004 ITDSP) N D(z) = Π (1-pkz-1) k=1 ▲pk is the error or perturbation resulting from the quantization of the filter coefficients 21.2. In caseof decimation by M.e. The anti-aliasing filter is LPF whose frequency response HLPF(ω) is given by HLPF(ω) = 1.Give the rounding errors for fixed and floating point arithmetic.3.What is a decimator? If the input to the decimator is x(n)={1. The error εR incurred due to rounding off floating point number is in the range -2E. D  Decimator 22.. (APR 2004 ITDSP) A number x represented by b bits which results in bR after being Rounded off. 20. otherwise. i. Interpolation is a process of increasing the sampling rate by a factor I.2}. up-sampling. most of its energy is contained in the low frequencies and hence can be coded with more bits then high frequencies. The spectrum X(ω) is assumed to be non-zero in the frequency interval 0≤|ω|≤π.2-bR/2)≤ εR ≤2E. the input spectrum X(ω) is band limited to the low pass frequency response.5.4.

When multiplication or addition is performed in digital filter.79) 2.. 24. errors are introduced due to their finite accuracy.4. I.0. which is called dead band.Decimation is a process of reducing the sampling rate by a factor D.What is meant by finite word length effects in digital filters? (Nov/Dec 2003)-ECE The digital implementation of the filter has finite accuracy.SIGNALS AND SYSTEMS 1. Therefore the result is quantized so that it can be represented by finite word register.5.e. This quantization error can create noise or oscillations in the output.(Pg no 2.0.What is dead band? (Nov/Dec 2004)-ECE In a limit cycle the amplitude of the output are confined to a range of value. Use convolution to find x(n) if X(z) is given by Nov/Dec 2008 CSE for ROC Refer book : Digital signal processing by Ramesh Babu .-1. Determine the causal signal x(n) having the Z transform Nov/Dec 2008 CSE X(z)= Refer book : Digital signal processing by Ramesh Babu . These effects are called finite word length effects.2} D=2 Output y(n) = {1.How can overflow limit cycles be eliminated? (Nov/Dec 2004)-ECE  Saturation Arithmetic  Scaling 25. These errors generate finite precision effects or finite word length effects.Determine whether the following signals are Linear . causal and stable (1) Y(n)=cos[x(n)] Nov/Dec 2008 CSE (2) Y(n)=x(-n+2) (3) Y(n)=x(2n) (4) Y(n)=x(n)+nx(n+1) Refer book : Digital signal processing by Ramesh Babu .66) 3. PART B UNIT-1 .(Pg no 1.3} 23.62) .3.2.(Pg no 2.-1. downsampling.Time Variant. the result is to be represented by finite word length (bits). x(n)={1. When numbers are represented in digital form.

0≤n≤4 0. and stable Refer book : Digital signal processing by Ramesh Babu .Nagoor kani .4.1} April/May2008CSE Refer book: Digital signal processing by A.(i)Determine the Z-transform of the signal x(n)=anu(n)-bnu(-n-1).find rxy and r yx for x={1. b>a and plot the ROC.185-1.cosWo(n) Nov/Dec 2007 CSE Refer book : Digital signal processing by Ramesh Babu . stable or unstable. (ii) Determine the IZT of X(z)=1 / [(1-z-1)(1-z-1)2] Nov/Dec 2007 CSE Refer book : Digital signal processing by A. y(n)-y(n-1)=x(n)+x(n-1) to inputs x(n)=u(n) and x(n)=2-n u(n).0. CSE Refer book : Digital signal processing by Ramesh Babu .3.79) 7.Nagoor kani . Test its stability.197) 9.A discrete-time system can be static or dynamic.(Pg no 23-24) 5.(Pg no 1.0.57) (ii) Find convolution of {5. Time invariant or time varying. .(i) Determine the response of the casual system.4.79) April/May2008 6.3.Find the response of the system if the input is {1.(Pg no 1.casual. -3≤n≤5 0 .2} and impulse response of the system is {1.2} and {1.2.2} April/May2008 CSE Refer book : Digital signal processing by Ramesh Babu .0. elsewhere and h (n)= 1.2}.(Pg no 1. (i) Check whether the system y(n)=ay(n-1)+x(n) is linear . (i) Compute the convolution y(n) of the signals x(n)= an. Examine the following system with respect to the properties also.6. (1) y(n) = cos [x(n)] (2) y(n)=x(-n+2) (3) y(n)=x(2n) (4) y(n)=x(n).3} and y={4. elsewhere Nov/Dec 2007 CSE 8.51-1. (Pg no 463) 10. shift variant. linear or nonlinear.1.3.2.4.(Pg no 1. causal or non causal.

3. Page number (207) (iii) Find the system function of the system described by y(n)-0.Nagoor kani .(Pg no 23-24) 14.find rxy and r yx for x={1.75y(n-1)+0.n<0 Refer book : Digital signal processing by Ramesh Babu .0.1.79) 15.20) (iii) prove the property that convolution in Z-domains multiplication in time domain April/May2008 IT Refer book : Digital signal processing by Ramesh Babu .-4} and h(n)={0.4. (i) Compute the z-transform and hence determine ROC of x(n) where (1/3) n (1/2) -n u(n).(Pg no 1.6.2}.79) . 2000.n ≥ 0 X (n) = u(n).2} and {1.(i) find the convolution and correlation for x(n)={0.3} and y={4. Refer book : Digital signal processing by Ramesh Babu . “Digital Signal Processing Principles.(Pg no 2.4. and stable Refer book : Digital signal processing by Ramesh Babu .2.(i) Check whether the system y(n)=ay(n-1)+x(n) is linear . April/May2008 CSE Refer book : Digital signal processing by Ramesh Babu .casual.3.0.1} April/May2008CSE Refer book: Digital signal processing by A. shift variant.2} April/May2008 CSE Refer book : Digital signal processing by Ramesh Babu .-2.5/[(1-0.0. PHI/Pearson Education.3.Refer John G Proakis and Dimtris G Manolakis.Find the response of the system if the input is {1.(Pg no 1.1. Page number (157) (ii) Find the steady state value given Y(z)={0.5}.3.77) 13. Algorithms and Application”.125y(n-2)=x(n)-x(n-1) and plot the poles and zeroes of 11. 3rd Edition. Algorithms and Application”.5.(Pg no 1.57) (ii) Find convolution of {5.(Pg no 1.2} and impulse response of the system is {1. PHI/Pearson Education.2.57) 12.2. 3rd Edition.(Pg no 1.75z-1)(1-z-1)]} Refer John G Proakis and Dimtris G Manolakis.1.79) (ii)Determine the Impulse response for the difference equation Y(n) + 3 y(n-1)+2y(n-2)=2x(n)-x(n-1) April/May2008 IT Refer book : Digital signal processing by Ramesh Babu .(Pg no 2.1.51-1. “Digital Signal Processing Principles.0. 2000.

65 (To find the impulse response h(n) and take z-transform.(i)Using Z-transform.(i) Determine the response of the casual system.(i)Determine the Z-transform of the signal x(n)=anu(n)-bnu(-n-1).43 to 4. PHI/Pearson Education. y(n)-y(n-1)=x(n)+x(n-1) to inputs x(n)=u(n) and x(n)=2-n u(n). (MAY 2006 ITDSP) Refer signals and systems by P. Examine the following system with respect to the properties also. elsewhere and h (n)= 1. (1) y(n) = cos [x(n)] (2) y(n)=x(-n+2) (3) y(n)=x(2n) (4) y(n)=x(n).(Pg no 1. Page number (157) (ii) Find the steady state value given Y(z)={0.16. (i) Compute the convolution y(n) of the signals x(n)= an. b>a and plot the ROC.A discrete-time system can be static or dynamic. Refer John G Proakis and Dimtris G Manolakis. elsewhere Nov/Dec 2007 CSE 17.12y(n-2+x9n-1)+x(n-2) to the input x(n)=nu(n).125y(n-2)=x(n)-x(n-1) and plot the poles and zeroes of H(z). page no:4. Algorithms and Application”. causal or non causal. page no:4. 3 rd Edition. Test its stability. PHI/Pearson Education. linear or nonlinear. page no:10. compute the response of the system y(n)=0. Page number (207) (iii) Find the system function of the system described by y(n)-0.45 .99 (ii)State and prove the properties of convolution sum.75y(n-1)+0. Algorithms and Application”. (MAY 2006 ECESS) Refer signals and systems by chitode.cosWo(n) Nov/Dec 2007 CSE Refer book : Digital signal processing by Ramesh Babu .75z-1)(1-z-1)]} Refer John G Proakis and Dimtris G Manolakis.) 20. Ramesh babu . (ii) Determine the IZT of X(z)=1 / [(1-z-1)(1-z-1)2] Nov/Dec 2007 CSE Refer book : Digital signal processing by A.7y(n-1)-0. “Digital Signal Processing Principles. stable or unstable. Time invariant or time varying. 0≤n≤4 0. (Pg no 463) 19.5/[(1-0. Is the system stable? Refer signals and systems by chitode. 2000.185-1.Nagoor kani . -3≤n≤5 0 .197) 18. “Digital Signal Processing Principles. 3 rd Edition. 2000.

Also give the illustration Refer signals and systems by chitode. Find DFT for {1.(Pg no 3.88) 3. y(n) = nx(n) (APRIL 2005 ITDSP) Refer John G Proakis and Dimtris G Manolakis. Nov/Dec 2008 CSE Refer book : Digital signal processing by Ramesh Babu .1.2}. Page number (6. page no:3.1. Find the convolution sum for the x(n) =(1/3)-n u(-n-1) and h(n)=u(n-1) Refer signals and systems by P.FAST FOURIER TRANSFORMS 1. Derive the key equation of radix 2 DIF FFT algorithm and draw the relevant flow graph taking the computation of an 8 point DFT for your illustration Nov/Dec 2008 CSE Refer book : Digital signal processing by Nagoor Kani .0.2.2)=x(n)+2x(n-1) when the input sequence is x(n)=4n u(n). Refer signals and systems by P.2. page no:67 c.(Pg no 215) 4.x2(n)={1. Ramesh babu . Check whether the following systems are linear or not 1.0. “Digital Signal Processing Principles.76. Convolve the following two sequences linearly x(n) and h(n) to get y(n). (NOV 2006 ECESS) Refer signals and systems by chitode. 3rd Edition. Ramesh babu .49 23.43 to 4. page no:4. PHI/Pearson Education.46) 2.3.47 to 4.21. y(n)-3y(n-1)-4y(n. (APRIL 2004 ITDSP) Refer John G Proakis and Dimtris G Manolakis.1} and h(n) ={2. 3rd Edition.Determine the sequence x3(n) corresponding to the circular convolution of the sequence x1(n)={2.1. (NOV2006 ECESS) Refer signals and systems by chitode.1}.Explain the properties of an LTI system. page no:4.3. 2000. State the difference between overlap save method and overlap Add method Nov/Dec 2008 CSE Refer book : Digital signal processing by Ramesh Babu . (NOV 2006 ECESS) Refer signals and systems by chitode.1} using FFT DIT butterfly algorithm and .4}. 2000.77 b. page no:3-2 to 3-7 22.a. Explain the properties of convolution.2.10) UNIT-II . Algorithms and Application”.45 24.State and prove the sampling theorem. y(n) = x2(n) 2. “Digital Signal Processing Principles. page no:3.1.(i)Determine the response of the system described by. Page number (67) 25. Nov/Dec 2008 CSE Refer book : Digital signal processing by Nagoor Kani .23 (ii)Write the importance of ROC in Z transform and state the relationship between Z transforms to Fourier transform.By means of DFT and IDFT .(Pg no 3. Also explain how reconstruction of original signal is done from the sampled signal.1. Algorithms and Application”. “Digital Signal Processing”. PHI/Pearson Education.(Pg no 226) 5. Page number (153) Refer S Poornachandra & B Sasikala. x(n)= {1.2. Compute the FFT of the sequence x(n)=n+1 where N=8 using the in place radix 2 decimation in frequency algorithm.

89) .0.(Pg no 4.5.0.0. PHI/Pearson Education. (MAY 2006 ITDSP) Refer John G Proakis and Dimtris G Manolakis. (i) Compute the 8 pt DFT of the sequence x(n)={0. prove that April/May2008 IT X1(n)x2(n)=1/N [Xt(k) X2(k)].0. (Pg no 749) (ii)Discuss the use of FFT algorithm in linear filtering.34) (ii) Find 8 Point DFT of x(n)=0.Page number (8.87) 14.1} using FFT-DIF method April/May2008 CSE (ii)Find DFT for {1.5. April/May2008 IT 11.4.plot the spectrum CSE April/May2008 Refer book : Digital signal processing by Ramesh Babu .(i) Discuss the properties of DFT.89) (ii) Determine the number of complex multiplication and additions involved in a Npoint Radix-2 and Radix-4 FFT algorithm. Algorithms and Application”. otherwise using Decimation-in-time FFT algorithm (APRIL 2005 ITDSP) Refer P. (i)Find IDFT for {1. Ramesh babu.0} using radix2 decimation in time and radix2 decimation in frequency algorithm. 4≤n≤7 April/May2008 IT Refer book : Digital signal processing by Ramesh Babu . “Signals and Systems”. Page number (456 & 465) 12.(Pg no 4. “Signals and Systems”.5. 3 rd Edition.Page number (8.0.0} using DIT FFT (NOV 2005 ITDSP) Refer P.3.1} (MAY 2006 ITDSP) Refer book : Digital signal processing by Ramesh Babu .B. Ramesh babu.0.0.2.0.Compute the eight point DFT of the sequence x(n)={ ½.Find the 8-pt DFT of the sequence x(n)={1.0} (APRIL 2005 ITDSP) Refer P. 2000.(Pg no 3.0..0.(Pg no 4.4.Page number (8. Refer book : Digital signal processing by S.Compute the 8 pt DFT of the sequence x(n)={0.0.32) 10. “Signals and Systems”.0.0≤n≤3 Using DIT FFT 0.Find the 8-pt DFT of the sequence x(n)= 1.5. Nov/Dec 2007 CSE Refer book : Digital signal processing by John G.sasikala. Follow exactly the corresponding signal flow graph and keep track of all the intermediate quantities by putting them on the diagram. Ramesh babu.30) 8.29) 7.(Pg no 4.0} using radix-2 DIT FFT Refer P.0. N Refer book : Digital signal processing by Ramesh Babu .5. “Signals and Systems”.17) 6.(i) if x(n) N pt DFT X(k) then.0.Poornachandra.0.5.Derive the equation for radix 4 FFT for N=4 and Draw the butterfly Diagram.5.(Pg no 447) 9.½. “Digital Signal Processing Principles. 0≤n≤7 0.58) 13.0.Proakis . Nov/Dec 2007 CSE Refer book : Digital signal processing by Ramesh Babu .5.½.½.1.5. Page number (8. Ramesh babu.3.

(APR 2004 ITDSP) & (MAY 2006 ECEDSP) Refer P.2.Determine the direct form I .1} and x(n)={1.(Pg no5.Determine 8 pt DFT of x (n)=1for -3≤n≤3 using DIT-FFT algorithm (APR 2004 ITDSP) Refer P. Ramesh babu.0} Refer P.80) 18. Page number (8.58) (ii)Find the output sequence y(n) if h(n)={1.By means of DFT and IDFT .4} using circular convolution (APR 2004 ITDSP) Refer P.Page number (8. Ramesh babu.Design a digital butter worth filter satisfying the constraints Nov/Dec 2008 CSE 0.65) 17.46) 4.70-8.58) 19.direct form II . Determine x1(n) in terms of x(n) (NOV 2004 ITDSP) Refer John G Proakis and Dimtris G Manolakis.1.If the DFT of X(k)is computed to obtain a sequence x1(n).Cascade and parallel structure for the system Y(n)=-0. PHI/Pearson Education. “Signals and Systems”.1}. Page number (456 & 465) UNIT-III .61) 3.25x(n-2) Nov/Dec 2008 CSE Refer book : Digital signal processing by Ramesh Babu . Ramesh babu.Page number (8.(Pg no5.1.2 ms April/May2008 CSE Refer book : Digital signal processing by A.2.87) 16.(Pg no5. determine the response of an FIR filter with impulse response h(n)={1. 2000.IIR FILTER DESIGN 1.Design a digital filter corresponding to an analog filter H(s)= using the impulse invariant method to work at a sampling frequency of 100 samples/sec Nov/Dec 2008 CSE Refer book : Digital signal processing by Ramesh Babu .7x(n)-0.707≤ ≤1 for 0 ≤w≤ ≤0. “Signals and Systems”.Let X(k) denote the N-point DFT of an N-point sequence x(n).0.n=0.15. “Signals and Systems”.2. (i)What is decimation in frequency algorithm? Write the similarities and differences between DIT and DIF algorithms.3}.2 to the input sequence x(n) ={1.What is the main drawback of impulse invariant method ?how is this overcome by bilinear transformation? Nov/Dec 2008 CSE Refer book : Digital signal processing by Ramesh Babu .1y(n-1)+0.2.40) 2. Page number (8. (i)Design digital filter with H(s) = using T=1sec.Page number (8. (ii)Design a digital filter using bilinear transform for H(s)=2/(s+1)(s+2)with cutoff frequency as 100 rad/sec and sampling time =1. “Signals and Systems”.(i)Determine the 8 point DFT of the sequence x(n)= {0. Ramesh babu.3.1.Nagoor kani . Ramesh babu.(Pg no5. 3rd Edition.1. Algorithms and Application”. (NOV 2005 ITDSP) Refer P.79) 5. (i) Realize the following filter using cascade and parallel form with .(Pg no 341) 6.1. “Digital Signal Processing Principles.20 for ≤w≤ With T=1 sec using bilinear transformation .0.72y(n-2)+0.realize the same in Direct form II Refer book : Digital signal processing by Ramesh Babu .0. “Signals and Systems”.

(Assume T=0.2y(n-2)+3x(n)+3.(Pg no 5.7x(n)-0. cascade and parallel realization for the following system Y(n)= -0. (APRIL 2005 ITDSP) Refer John G Proakis and Dimtris G Manolakis.Determine the Direct form II realization for the following system: y(n)=-0. 2000. PHI/Pearson Education. Algorithms and Application”.5 KHz and an attenuation of 40 dB at 3.2п by use of bilinear transformation.direct form –I structure 1+z-1 +z -2+ 5z-3 ( 1+Z-1)(1+2Z-1+4Z-2) ( ii) Find H(s) for a 3 rd order low pass butter worth filter April/May2008 CSE Refer book : Digital signal processing by Ramesh Babu .1)/[(s+0.8) 7.(i) Convert the analog filter system functionHa(s)={(s+0.35∏ ≤|w| ≤∏ Use Bilinear Transformation Refer book : Digital signal processing by Ramesh Babu .(i) Derive bilinear transformation for an analog filter with system function H(s) =b / (s+a) Refer book: Digital signal processing by John G.1y(n-1+0. Design a low pass Butterworth filter that has a 3 dB cut off frequency of 1. “Digital Signal Processing Principles.(Pg no 5.42) (ii) Determine the order of Cheybshev filter that meets the following specifications (1) 1 dB ripple in the pass band 0≤|w| ≤ 0.9b) 13.68) (ii) Discuss the limitation of designing an IIR filter using impulse invariant method.27) 11. Page number (675) 12. Direct Form II.Nagoor kani .6x(n-1)+0. Page number (601-7.Proakis . .(Pg no 5. (Pg no 330) 9.1)2+9]} into a digital IIR filter using impulse invariance method.14) 10. 3rd Edition. PHI/Pearson Education.(Pg no 5.6x(n-2) Refer book : Digital signal processing by Ramesh Babu . “Digital Signal Processing Principles. Algorithms and Application”.252x(n-2).(i) Obtain the Direct Form I. 3 rd Edition.1sec) (APR 2006 ECEDSP) Refer John G Proakis and Dimtris G Manolakis.1y(n-1)+0.(Pg no 5. (i) Use the Impulse invariance method to design a digital filter from an analog prototype that has a system function April/May2008 IT Ha(s)=s+a/((s+a)2 +b2 ) Refer book : Digital signal processing by Ramesh Babu .(Pg no 676-679) (ii)Design a single pole low pass digital IIR filter with -3 db bandwidth of 0.Explain the method of design of IIR filters using bilinear transform method. Nov/Dec 2007 CSE Refer book : Digital signal processing by A. Nov/Dec 2007 CSE 8.72y(n-2)+0.3 b (2) Atleast 60 dB attrnuation in the stop band 0. 2000.0 kHz April/May2008 IT Refer book : Digital signal processing by Ramesh Babu .

Page number (506 &531) 15.37) 16. PHI/Pearson Education. 2000.Obtain cascade form structure of the following system: y(x)=-0. Page number (209) UNIT-IV .75y(n-1)+0. “Digital Signal Processing Principles. PHI/Pearson Education. Page number (601-7. Page number (601-7. “Digital Signal Processing Principles. 3 rd Edition. 3 rd Edition.(APRIL 2005 ITDSP) Refer John G Proakis and Dimtris G Manolakis.Explain the type I frequency sampling method of designing an FIR filter. PHI/Pearson Education. 2000. page no:3.Find the output of an LTI system if the input is x(n)=(n+2) for 0≤n≤3 and h(n)=a nu(n) for all n (APR 2004 ITDSP) Refer signals and systems by P. Algorithms and Application”.9c) 18.FIR FILTER DESIGN 1. 2000. PHI/Pearson Education.Realize the same in Direct form II Refer book : Digital signal processing by Nagoor Kani . PHI/Pearson Education. 3 rd Edition. Ramesh babu . Page number (676-8. Determine its system function (NOV 2005 ITDSP) Refer John G Proakis and Dimtris G Manolakis. 2000.6x(n-2) (APR 2004 ITDSP) Refer John G Proakis and Dimtris G Manolakis. “Digital Signal Processing Principles.2y(n-2)+3x(n)+3.6x(n-1)+0.125y(n-2)=x(n)+0. 3rd Edition. 2000.(Pg no 78) 3.Obtain direct form and cascade form realizations for the transfer function of the system given by Nov/Dec 2008 CSE Refer book : Digital signal processing by Nagoor Kani . Algorithms and Application”.38 17. 3 rd Edition.3) 14.Consider the system described by y(n)-0.5Z-2) (APR 2004 ITDSP) Refer John G Proakis and Dimtris G Manolakis.Verify the Stability and causality of a system with H(z)=(3-4Z-1)/(1+3. . Algorithms and Application”.33x(n-1). “Digital Signal Processing Principles.3. Algorithms and Application”.Explain the following terms briefly: (i)Frequency sampling structures (ii)Lattice structure for IIR filter (NOV 2005 ITDSP) Refer John G Proakis and Dimtris G Manolakis.Design a FIR linear phase digital filter approximating the ideal frequency response Nov/Dec 2008 CSE With T=1 Sec using bilinear transformation .(Pg no 367) 2. “Digital Signal Processing Principles.1y(n-1)+0. Algorithms and Application”.5Z-1+1.

Describe the window sequences generally used and compare their properties.….82) 4.(Pg no 292-295) 7. Nov/Dec 2007 Refer book: Digital signal processing by John G. Design a LPF for the following response . Nov/Dec 2007 CSE 8.28) 5.Explain how a linear phase FIR filter can be used using window method.using hamming window with N=7 April/May2008 CSE 6. Nov/Dec 2007 CSE Refer book : Digital signal processing by A.(Pg no 630-632) (ii)A low pass filter has the desired response as given below e-i3w.Compare the frequency domain characteristics of various window functions .Nov/Dec 2008 CSE Refer book : Digital signal processing by Ramesh Babu .(i) For FIR linear phase Digital filter approximating the ideal frequency response Hd(w) = 1 ≤|w| ≤∏ /6 0 ∏ /6≤ |w| ≤∏ Determine the coefficients of a 5 tap filter using rectangular Window jw .(Pg no6. 0≤w<∏/2 Hd(e )= 0.(I) Explain the type 1 design of FIR filter using frequency sampling technique. Nov/Dec 2007 CSE Refer book : Digital signal processing by A. (i) Prove that an FIR filter has linear phase if the unit sample response satisfies the condition h(n)= ±h(M-1-n).Nagoor kani .1.Proakis .(Pg no6. Nov/Dec 2008 CSE Refer book : Digital signal processing by Ramesh Babu . ∏/2≤<∏ Determine the filter coefficients h(n) for M=7 using frequency sampling method. Also discuss symmetric and antisymmetric cases of FIR filter. n=0.M-1. (Pg no 630-632) (ii) Explain the need for the use of window sequences in the design of FIR filter.Nagoor kani .

Algorithms and Application”.Refer book : Digital signal processing by A.2 & 8.(i) Determine the coefficient h(n) of a linear phase FIR filter of length M=5 which has symmetric unit sample response and frequency response Hr(k)=1 for k=0.4 for k=4 0 for k=5.2.2.2. 2000.(Pg no 415 (ii) Determine the unit sample response h(n) of a linear phase FIR filter of Length M=4 for which the frequency response at w=0 and w= ∏/2 is given as Hr(0) . 3 rd Edition.Nagoor kani . Design linear HPF using Hanning Window with N=9 H(w) =1 -п to Wc and Wc to п =0 otherwise April/May2008 IT Refer book : Digital signal processing by A. Algorithms and Application”.Nagoor kani . 3rd Edition. Page number (FIR-118) UNIT-V .Explain the steps involved in the design of FIR Linear phase filter using window method.Nagoor kani . PHI/Pearson Education.3 0. 7 April/May2008 IT(NOV 2005 ITDSP) Refer book : Digital signal processing by A.(Pg no 310) 9.3) 13. 3rd Edition. “Digital Signal Processing Principles. Page number (8.FINITE WORD LENGTH EFFECTS 1.(Pg no 301) 11.172) . 2000. (APR 2005 ITDSP) Refer John G Proakis and Dimtris G Manolakis.Draw the circuit diagram of sample and hold circuit and explain its operation Nov/Dec 2008 CSE/ Nov/Dec 2007 CSE Refer book : Digital signal processing by Ramesh Babu . 2000. “Digital Signal Processing Principles. PHI/Pearson Education. (NOV 2004 ITDSP) & ( NOV 2005 ITDSP) Refer John G Proakis and Dimtris G Manolakis.(Pg no 308) m-1 (ii) Show that the equation ∑ h(n)=sin (wj-wn)=0.(Pg no1.Hr(∏/2) =1/2 April/May2008 IT Refer book : Digital signal processing by A. “Digital Signal Processing Principles.is satisfied for a linear phase FIR filter n=0 of length 9 April/May2008 IT 10.Nagoor kani . “Digital Signal Processing”.Explain in detail about frequency sampling method of designing an FIR filter.(i)What are the issues in designing FIR filter using window method? Refer John G Proakis and Dimtris G Manolakis.1. Page number (8. PHI/Pearson Education. 6. Algorithms and Application”.2) (ii)An FIR filter is given by y(n)=2x(n)+(4/5)x(n-1)+(3/2)x(n-2)+(2/3)x(n-3) find the lattice structure coefficients (APR 2004 ITDSP) Refer S Poornachandra & B Sasikala. Page number (630) 12.

Describe the quantization errors that occur in rounding and truncation in two’s complement.(Pg no 564) m 8.2) 12.Proakis . PHI/Pearson Education.(Pg no 8.(Pg no 415) 9.7. H1(z) = 1/ (1-0.Nagoor kani . The input of the system y(n)=0.(Pg no 423) 3.Assume x(0) =0.the system function of the individual section are H1(z)=1/(1-0.Draw the product quantization noise model of the system and determine the overall output noise power April/May2008 IT Refer book : Digital signal processing by A.(Pg no10. “Digital Signal Processing .99y(n-1)+x(n) is applied to an ADC . 2000.2.(Pg no 412) 9.What is vocoder? Explain with a block diagram Nov/Dec 2008 CSE/ Nov/Dec 2007 CSE Refer book : Digital signal processing by Ramesh Babu .24) 7.(Pg no 423-426) 11.8z-1) Nov/Dec 2007 CSE Refer book: Digital signal processing by Ramesh Babu. 3 rd Edition.Discuss the limit cycle in Digital filters Nov/Dec 2008 CSE Refer book : Digital signal processing by Nagoor Kani .7.9-7.(i) Explain clearly the downsampling and up sampling in multirate signal processing.A cascade Realization of the first order digital filter is shown below . Nov/Dec 2007 CSE Refer book : Digital signal processing by John G.1 &7.75 and y(-1)=0 Refer book : Digital signal processing by A. Explain product quantization and prove бerr2 =∑ б2oi April/May2008 IT i=1 Refer book : Digital signal processing by A.1) 5.Nagoor kani . Determine the overall output noise power. (ii) Discuss about quantization noise and derive the equation for finding quantization noise power. April/May2008CSE Refer book : Digital signal processing by Ramesh Babu.7) (ii) Discuss about multirate Signal processing April/May 2008 CSE Refer book : Digital signal processing by Ramesh Babu . Algorithms and Application”. Page number(7.14) 6. (APRIL 2005 ITDSP) Refer John G Proakis and Dimtris G Manolakis. Explain the following terms briefly: (i)Perturbation error (ii)Limit cycles (NOV 2005 ITDSP) Refer John G Proakis and Dimtris G Manolakis.(Pg no 420) 4.9z-1 ) and H2(z) =1/(1-0. (i) Explain how the speech compression is achieved . (Pg no 7.Nagoor kani .8z-1) . Two first order low pass filter whose system functions are given below are connected in cascade. (i) Show dead band effect on y(n) = .9z-1) and H2(z) = 1/ (1-0. “Digital Signal Processing Principles.what is the power produced by the quantization noise at the output of the filter if the input is quantized to 8 bits Nov/Dec 2008 CSE Refer book : Digital signal processing by Nagoor Kani .95 y(n-1)+x(n) system restricted to 4 bits .(Pg no 7.

(i) Derive the spectrum of the output signal for a decimator (ii) Find and sketch a two fold expanded signal y(n) for the input (APR 2004 ITDSP) &(NOV 2004 ITDSP) Refer John G Proakis and Dimtris G Manolakis. PHI/Pearson Education. “Digital Signal Processing Principles. Algorithms and Application”. 2000. PHI/Pearson Education. With examples illustrate (i) Fixed point addition (ii) Floating point multiplication (iii) Truncation (iv) Rounding. Write applications of multirate signal processing in Musical sound processing (NOV 2004 ITDSP) Refer John G Proakis and Dimtris G Manolakis. Page number(831-833) 13. Page number (788) 14. 3 rd Edition. PHI/Pearson Education. PHI/Pearson Education. Page number (784-790) (ii)Explain subband coding of speech signal (NOV 2003 ITDSP) & (NOV 2004 ITDSP) & (NOV 2005 ITDSP) Refer John G Proakis and Dimtris G Manolakis. Algorithms and Application”. 2000. PHI/Pearson Education. Page number (12. 2000. Algorithms and Application”. 3 rd Edition. 3 rd Edition. Describe a single echo filter using in musical sound processing. 2000. Page number (952) 16. (NOV 2004 ITDSP) Refer John G Proakis and Dimtris G Manolakis.(i)Propose a scheme for sampling rate conversion by a rational factor I/D. 2000. PHI/Pearson Education. “Digital Signal Processing Principles. 3 rd Edition. 3 rd Edition. (APRIL 2004 ITDSP) Refer John G Proakis and Dimtris G Manolakis. PHI/Pearson Education. 2000.3) STAFF INCHARGE HOD/ECE . Page number (7.5.(APR 2005 ITDSP) & (NOV 2003 ITDSP) Refer John G Proakis and Dimtris G Manolakis. 3 rd Edition. “Digital Signal Processing Principles. Algorithms and Application”. “Digital Signal Processing Principles. Page number (790) 15. “Digital Signal Processing Principles. 3 rd Edition. Algorithms and Application”. Algorithms and Application”.Principles. “Digital Signal Processing Principles. 2000. Algorithms and Application”.5) 17.

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